git.asterisk.org Git - asterisk/asterisk.git/log

dsp_process was enhanced to work with alaw and ulaw in addition to slin.
noticed that some functions could be refactored here it is.

Reported by: irroot
Tested by: irroot, mnicholson
Review: https://reviewboard.asterisk.org/r/1304/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329432 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Remove lastmsgssent from sip it has not been working since 1.6
Clean up the return values to be consistant not currently used
Add doxygen returns
MWI Event is sent on Register

(closes issue ASTERISK-17866)
Reported by: one47
Tested by: irroot, mvanbaak
Review: https://reviewboard.asterisk.org/r/1172/

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r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011) | 1 line

Make use less redundant loop construct for iterating over hints.
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  r329331 | rmudgett | 2011-07-22 15:43:07 -0500 (Fri, 22 Jul 2011) | 55 lines

  Merged revisions 329299 via svnmerge from
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    r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) | 48 lines

    Deadlocks dealing with dialplan hints during reload.

    There are two remaining different deadlocks reported dealing with dialplan
    hints.

    The deadlock in ASTERISK-17666 is caused by invalid locking order in
    ast_remove_hint().  The hints container must be locked before the hint
    object.

    The deadlock in ASTERISK-17760 is caused by a catch-22 situation in
    handle_statechange().  The deadlock is caused by not having the conlock
    before calling the watcher callbacks.  Unfortunately, having that lock
    causes a different deadlock as reported in ASTERISK-16961.

    * Fixed ast_remove_hint() locking order.

    * Made handle_statechange() no longer call the watcher callbacks holding
    any locks that matter.

    * Made hint ao2 destructor do the watcher callbacks for extension
    deactivation to guarantee that they get called.

    * Fixed hint reference leak in ast_add_hint() if the callback container
    constructor failed.

    * Fixed hint reference leak in complete_core_show_hint() for every hint it
    found for CLI tab completion.

    * Adjusted locking in ast_merge_contexts_and_delete() for safety.

    * Added context_merge_lock to prevent ast_merge_contexts_and_delete() and
    handle_statechange() from interfering with each other.

    * Fixed ast_change_hint() not taking into account that the extension is
    used for the hash key.

    (closes issue ASTERISK-17666)
    Reported by: irroot
    Tested by: irroot
    JIRA SWP-3318

    (closes issue ASTERISK-17760)
    Reported by: Byron Clark
    Tested by: irroot
    JIRA SWP-3393

    Review: https://reviewboard.asterisk.org/r/1313/
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r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines

s/1.10/10.0/
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  r329204 | rmudgett | 2011-07-21 13:05:18 -0500 (Thu, 21 Jul 2011) | 13 lines

  Merged revisions 329203 via svnmerge from
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    r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) | 6 lines

    Document parkinglot in chan_dahdi.conf.sample.

    * Document existing feature in chan_dahdi.conf.sample.

    * Remove some dead code related to the parkinglot option.
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  r329200 | rmudgett | 2011-07-21 12:32:02 -0500 (Thu, 21 Jul 2011) | 24 lines

  Merged revisions 329199 via svnmerge from
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    r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) | 17 lines

    Update PickupChan documentation.

    The PickupChan uses the ampersand as the argument separator.
    Was documented as:
    PickupChan(channel[,channel2[,...][,options]])

    Fixed documentation to:
    PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])

    This is a continuation of ASTERISK-17494 for v1.8 and later.

    (closes issue ASTERISK-18144)
    Reported by: Erik Smith
    Patches:
          pickupchan_ducumentation-v2.patch (License #6263) patch uploaded by Erik Smith
    Tested by: Erik Smith
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  r329145 | rmudgett | 2011-07-21 11:52:17 -0500 (Thu, 21 Jul 2011) | 16 lines

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    r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines

    Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked!

    This appears to be a leftover from when ast_channel was converted to ao2
    objects.

    Simply removed the extraneous unlock.

    (closes issue ASTERISK-17772)
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Fix UPGRADE.txt files for Asterisk 10.

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Remove another 2.0 property.

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Fix merge properties to reflect Asterisk 10 branch

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  r329055 | pabelanger | 2011-07-20 17:27:50 -0400 (Wed, 20 Jul 2011) | 9 lines

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    r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, 20 Jul 2011) | 2 lines

    Asterisk now requires libpri 1.4.11+ for PRI support.
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  r328992 | twilson | 2011-07-20 15:18:25 -0500 (Wed, 20 Jul 2011) | 12 lines

  Merged revisions 328987 via svnmerge from
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    r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) | 5 lines

    We can't guarantee an eth0 is present

    FreeBSD test fails on this case presumably because there is no eth0 on the test
    machine. Better to just remove this test for now.
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  r328936 | kmoore | 2011-07-20 14:01:37 -0500 (Wed, 20 Jul 2011) | 15 lines

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    r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | 8 lines

    Inband DTMF regression

    The functionality of inband DTMF in chan_sip relied upon
    ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
    ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
    documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
    never inband.  This fixes the regression introduced in revision 328823.
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  r328879 | kpfleming | 2011-07-19 16:31:16 -0500 (Tue, 19 Jul 2011) | 23 lines

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    r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul 2011) | 17 lines

    Revert partial attempt at handling pathnames with spaces.

    Revision 299794 attempted to improve the build system to be able to handle
    pathnames (primarily DESTDIR) with spaces in them, since this is common on
    some platforms (including Mac OSX). Unfortunately, the changes were incomplete
    and did not actually provide the desired behavior, and as a side effect the
    functionality that ensured that stale headers in the Asterisk 'include' directory
    were removed got broken. In addition, the check for stale (and possibly
    incompatible) modules in the Asterisk 'modules' directory also got broken, and
    would never report any stale modules. Users upgrading to this version or later
    versions would then see unexpected module load errors.

    Since there are few users who actually want to install Asterisk into paths
    that contain spaces, and a proper fix for the build system would take many hours,
    the best solution for now is to just revert the partial solution.
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Edit the merge properties to match their names.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328880 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fix properties after twilson's change so merges still work

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  r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines

  Merged revisions 328823 via svnmerge from
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    r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines

    RTP bridge away with inband DTMF and feature detection

    When deciding whether Asterisk was allowed to bridge the call away from the
    core, chan_sip did not take into account the usage of features on dialed
    channels that require monitoring of DTMF on channels utilizing inband DTMF.
    This would cause Asterisk to allow the call to be locally or remotely bridged,
    preventing access to the data required to detect activations of such features.

    (closes 17237)
    Review: https://reviewboard.asterisk.org/r/1302/
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  r328771 | kmoore | 2011-07-19 10:46:54 -0500 (Tue, 19 Jul 2011) | 18 lines

  Merged revisions 328770 via svnmerge from
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    r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | 11 lines

    MeetMe requests a PIN twice in some circumstances

    If a call to MeetMe includes both the dynamic(D) and always request PIN(P)
    options, MeetMe will ask for the PIN two times: once for creating the
    conference and once for entering the conference.  This behavior was introduced
    in rev 311616 when adding the CONFFLAG_ALWAYSPROMPT option to the logic branch
    controlling PIN entry for joining a conference.

    (closes AST-601)
    Review: https://reviewboard.asterisk.org/r/1305/
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  r328717 | twilson | 2011-07-18 20:55:32 -0500 (Mon, 18 Jul 2011) | 14 lines

  Merged revisions 328716 via svnmerge from
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    r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) | 7 lines

    Make AST_LIST_REMOVE safer

    AST_LIST_REMOVE shouldn't modify the element passed in if it isn't found. This
    commit also adds linked list unit tests.

    Review: https://reviewboard.asterisk.org/r/1321/
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  r328664 | markm | 2011-07-18 16:50:13 -0400 (Mon, 18 Jul 2011) | 15 lines

  Merged revisions 328663 via svnmerge from
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    r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | 9 lines

    app_dial may double free a channel datastore

    When starting a call with originate, and having the callee channel run Bridge() on pickup, we will double free the dialed_interface_info datastore, causing a crash.  Make sure to check if the datastore still exists before trying to free it.

    (closes issue ASTERISK-17917)
    Reported by: Mark Murawski
    Tested by: Mark Murawski
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  r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines

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    r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines

    If the sip private structure is null, sip_setoption() will defref the null pointer and crash.

    Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure.  But this will fix a crash.

    (closes issue ASTERISK-17909)
    Reported by: Mark Murawski
    Tested by: Mark Murawski
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  r328609 | markm | 2011-07-18 08:37:53 -0400 (Mon, 18 Jul 2011) | 15 lines

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    r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | 8 lines

    Fixed invalid read and null pointer deref on asterisk shutdown.

    In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash.

    (closes issue ASTERISK-17927)
    Reported by: Mark Murawski
    Tested by: Mark Murawski, Kinsey Moore, Tilghman Lesher
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  r328541 | tilghman | 2011-07-18 02:11:26 -0500 (Mon, 18 Jul 2011) | 9 lines

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    r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 Jul 2011) | 2 lines

    Typo
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  r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, 15 Jul 2011) | 13 lines

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    r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 lines

    small gk processing fixes:
    - decrease for 1 second registration ttl for very low expirations (some
      providers expire few earlier than TTL)
    - delete rrq and registration expire timers on URQ received as we make
      new registration.
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  r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, 15 Jul 2011) | 2 lines

  delete unproperly changed svn props
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r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) | 1 line

Build app_macro by default because things depend on it.
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  r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) | 2 lines

  Update UPGRADE.txt and CHANGES files.
  Update documentation files stating that deprecated modules are no longer built by default.
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Add SLA to skinny.

Adds sublines to skinny lines. Each subline can be attached to an
SLA station/trunk combo. Includes the following functionality:

Callid is persistent for both in/out calls on all skinny devices.
Can join, hold, resume.
All sublines appear under a single line button.

See: https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for doc.

(closes issue ASTERISK-17947)

Review: https://reviewboard.asterisk.org/r/1239/

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r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines

Make hint watcher callback take const strings for context and exten parameters.
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  r328317 | rmudgett | 2011-07-14 18:28:49 -0500 (Thu, 14 Jul 2011) | 13 lines

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    r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) | 6 lines

    Missing SIP pvt and channel unlock in sip_set_rtp_peer().

    Regression introduced by -r326144.

    Add missing SIP pvt and channel unlock in sip_set_rtp_peer().
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  r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines

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    r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines

    Introduce <support_level> tags in MODULEINFO.
    This change introduces MODULEINFO into many modules in Asterisk in order to show
    the community support level for those modules. This is used by changes committed
    to menuselect by Russell Bryant recently (r917 in menuselect). More information about
    the support level types and what they mean is available on the wiki at
    https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States
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  r328207 | jrose | 2011-07-14 14:45:18 -0500 (Thu, 14 Jul 2011) | 13 lines

  Merged revisions 328205 via svnmerge from
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    r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | 6 lines

    Monitor application arguments requirements fixed.

    Monitor was requiring options in spite of no individual option on Monitor being required.

    Review: https://reviewboard.asterisk.org/r/1320/
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  r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines

  tune the v21 preamble detector to properly detect the desired sequence of hits
  and misses
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  r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines

  Preserve sample rate quality of wideband mixmonitor recordings.

  MixMonitor has the ability to record in any file format Asterisk supports,
  but the quality of wideband audio is not preserved.  This is because
  regardless of the sample rate the call is being recorded in, the audio
  is always downsampled to 8khz and then upsampled to whatever wideband
  format it is being written as.  This patch resolves this by requesting
  the audio from the audiohook in the signed linear format closest to the
  sample rate of the format we are writing.  This fix is only possible for
  Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband
  audio.

  Review: https://reviewboard.asterisk.org/r/1314/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328121 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add UPGRADE-1.10.txt file from UPGRADE.txt.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328079 65c4cc65-6c06-0410-ace0-fbb531ad65f3

set 1.10 merge properties

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328076 65c4cc65-6c06-0410-ace0-fbb531ad65f3

remove 1.8 merge properties

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328075 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 328014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) | 1 line

Add ATXFER_NULL_TECH note in features.conf.sample.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328016 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327950 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul 2011) | 14 lines

  Correct double-free situation in manager output processing.

  The process_output() function calls ast_str_append() and xml_translate() on its
  'out' parameter, which is a pointer to an ast_str buffer. If either of these
  functions need to reallocate the ast_str so it will have more space, they will
  free the existing buffer and allocate a new one, returning the address of the
  new one. However, because process_output only receives a pointer to the ast_str,
  not a pointer to its caller's variable holding the pointer, if the original
  ast_str is freed, the caller will not know, and will continue to use it (and
  later attempt to free it).

  (reported by jkroon on #asterisk-dev)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327953 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327890 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue, 12 Jul 2011) | 2 lines

search in the current context for 'a' and 'o' instead of 'default'
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327891 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) | 1 line

Fix uninstall target, so that modules dir gets cleared again.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327889 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines

  Added additional checks for mailbox / password beginning with '*' character

  A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated.  The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character.

  (closes issue ASTERISK-17443)
  Reported by: Kevin Scott Adams
  Tested by: Matt Jordan

  Review: https://reviewboard.asterisk.org/r/1316/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327793 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) | 14 lines

  Use 'printf' (POSIX issue 4) instead of 'echo -n', for portability.

  The problem with using 'echo -n' is that it is not portable.  While BSD systems
  required that the '-n' option be removed and interpreted, System V required
  that all strings should be echoed with no interpretation of options.  This
  fundamental difference of behavior means that it is never possible to use the
  '-n' flag to echo in tests which are meant to be portable.

  In this case, on Mac OS X 10.6, the /bin/sh shell builtin 'echo' uses the
  System V semantics of the command, and thus the SHELL test failed on that
  platform.

  http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327794 65c4cc65-6c06-0410-ace0-fbb531ad65f3

do v21 detection instead of CED detection for the fax gateway

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327769 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Send video update frame to new video source in follow_talker correctly.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327749 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Segfault on shutdown when confbridge is active

When undergoing a shutdown and channels are kicked out of a bridge, a segfault
occurs because ConfBridge tries to play sounds on the bridge after the
underlying channels have been blown away due to the shutdown.

(closes ASTERISK-18040)
Review: https://reviewboard.asterisk.org/r/1283/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327748 65c4cc65-6c06-0410-ace0-fbb531ad65f3

use printf instead of echo -n in substitution test

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327684 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) | 9 lines

  Update chan_gtalk to work with changed GMail-based calls

  The messages sent by the GMail client have changed, but include the
  old-style messages as well. This patch checks for this case and
  uses the old-style offer.

  (closes issue ASTERISK-18084)
  Review: https://reviewboard.asterisk.org/r/1312/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327683 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Updates follow_talker video_mode in confbridge application.

follow_talker mode originally echoed the same video stream
to all participants. As the primary talker switched around, the
video stream would result in the talker seeing themselves. Now
the primary talker sees the last person who was talking rather than
themselves.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327640 65c4cc65-6c06-0410-ace0-fbb531ad65f3

renamed fax_gateway_send_ced() to fax_gateway_request_t38()

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327598 65c4cc65-6c06-0410-ace0-fbb531ad65f3

actually do something with the ced timeout, also added more debug output

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327570 65c4cc65-6c06-0410-ace0-fbb531ad65f3

write silence on the channel during t.38 negotiation

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327514 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul 2011) | 2 lines

reset our buffer each iteration when doing variable substitution
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327513 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Delay sending an CED tone generated T.38 reinvite to give the CED tone
generating party time to send its own T.38 reinvite.

Also don't forward frames through the gateway if we are negotiating T.38.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327511 65c4cc65-6c06-0410-ace0-fbb531ad65f3

fixed wording in a comment

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327469 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) | 5 lines

  fix building the Debian armhf (HardFloat) port

  Fixes http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385
  (Missing pthreads)
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327413 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Full T.38 handshaking and fax detection

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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327359 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Add .o files to svn:ignore property, since it's only ignored if locally configured to do so.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327246 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327211 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327212 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds entry in UPDATES.txt for removal of formats/format_sln16.c. Fixes typo in CHANGES as well.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327168 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Updates CHANGES log to reflect new slinear read/write file interpreters.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327148 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Support for writing and reading raw slin files 8khz-192khz.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327137 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Moves celt and silk format attribute files into res folder.

It was inconsistent to have the silk and celt format attribute
modules in the format file interpreter folder.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327116 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327107 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
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.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327047 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 327044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08 Jul 2011) | 2 lines

Resolve some set-but-unused-variable warnings.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327045 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326985 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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().
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327000 65c4cc65-6c06-0410-ace0-fbb531ad65f3

I think reviewboard broke this. The whole file was doubled.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326943 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds the format_attr_celt file which was also missing from the CELT pass through patch.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326904 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds missing celt.h file from celt pass-through support patch.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes spelling errors in CHANGES as well as adding a few entries for CELT and confbridge.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326856 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Adds pass-through support for codec CELT.

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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326830 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326842 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Updates confbridge.conf video documentation and adds dtmf action for releasing video src.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326782 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Use older functions out of deference to older distros

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326750 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326694 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

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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
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326684 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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

live_ast: valgrind: run asterisk under valgrind

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/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Replace Berkeley DB with SQLite 3

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.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes newlines from being stripped from out of dialog sip MESSAGES.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326544 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326484 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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.
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326485 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326412 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prompt conversion script

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)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326368 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326291 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326321 65c4cc65-6c06-0410-ace0-fbb531ad65f3

New feature: AMI Action FilterAdd

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

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326267 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Merged revisions 326209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  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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Change CHANGES move the commits to the right place
r296249 r318141 Application changes

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Change CHANGES move the commits to the right place in the file missed in review

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326056 65c4cc65-6c06-0410-ace0-fbb531ad65f3

updated irroots info for the authors section

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326006 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Fixes warning message caused by confbridge playback chan not being answered.

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325936 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Video support for ConfBridge.

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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Fax gateway functionality (i.e. translating between a T.30 terminal and a T.38
terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the
dialplan.

Big thanks to irroot for porting this code to use the framehooks api.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

copy all flags on asterisk frames instead of just the timing flag

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Merged revisions 325740 via svnmerge from
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  r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | 7 lines

  chan_sip: cleanup from the introduction of ast_str

  Remove the length field from sip_req and sip_pkt in chan_sip since they are
  redundant (ast_str holds its own length) and refactor the necessary functions.

  Review: https://reviewboard.asterisk.org/r/1281/
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  r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29 Jun 2011) | 6 lines

  Fixes timerfd locking issue.

  (closes ASTERISK-17867, ASTERISK-17415)
  Patches:
       fix uploaded by kobaz
  Review: https://reviewboard.asterisk.org/r/1255/
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