Multiple revisions 350555,350571

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  r350555 | rmudgett | 2012-01-13 11:12:51 -0600 (Fri, 13 Jan 2012) | 12 lines
  
  Add missing CEL logging fields to various CEL backends.
  
  * Add missing eventextra to cel_psql.c and cel_odbc.c.
  
  * Add missing PeerAccount and EventExtra to cel_manager.c.
  
  * Add missing userdeftype support for cel_custom.conf.sample and
  cel_sqlite3_custom.conf.sample.
  
  (closes issue ASTERISK-17190)
  Reported by: Bryant Zimmerman
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  r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 Jan 2012) | 8 lines
  
  Use compatible names for event extra data for various CEL backends.
  
  * Change eventextra to extra in cel_psql.c and cel_odbc.c.
  
  * Change EventExtra to Extra in cel_manager.c.
  
  (issue ASTERISK-17190)
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Previously, realtime queues could be loaded without defining the queue member
table.  This allowed for queue members to be dynamic, while the realtime
queue definitions could exist in some backing storage.  Revision 342223 broke
this when it changed the return value for realtime_multientry to return NULL
when no results are returned.  Previously, an empty ast_config object was
expected.

(closes issue ASTERISK-19170)
Reported by: Rene Mendoza
Tested by: Rene Mendoza
Patches: 
  rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283)
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This patch addresses two issues in ConfBridge and the channel bridge layer:
1. It fixes a race condition wherein the bridge channel could be hung up
2. It removes the deadlock avoidance from the bridging layer and makes the
   bridge_pvt an ao2 ref counted object

Patch by David Vossel (mjordan was merely the commit monkey)

(issue ASTERISK-18988)
(closes issue ASTERISK-18885)
Reported by: Dmitry Melekhov
Tested by: Matt Jordan
Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628)

(closes issue ASTERISK-19100)
Reported by: Matt Jordan
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1654/
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(closes issue ASTERISK-17940)
Reporter: Nic Colledge
Patches:
	features_18.patch uploaded by Nic Colledge (license 6245)
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(closes issue ASTERISK-19180)
Reported by: Corey Farrell
Patches:
      asterisk_cel_noevent_varset.diff (license #5909) patch uploaded by Corey Farrell
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Like Dial and Queue, FollowMe needs to deal with
AST_CONTROL_CONNECTED_LINE information so when the parties are initially
bridged, the connected line information will be correct.

* Added the 'I' option just like the app_dial and app_queue 'I' option.

* Made 'N' option ignored if the call is already answered.

(closes issue ASTERISK-18969)
Reported by: rmudgett
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1656/
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Initially, support was left in for the old style of searching, even
though it wasn't actually used. In the case of name_len != 0, the
OBJ_KEY flag isn't passed because we aren't matching on a full key
and therefor can't use the hash function to optimize. The code left
in to support the old way of searching unfortunately treated a prefix
search like this as though an ast_channel struct was passed as an arg
and caused a crash.

This patch also adds needed parentheses around some matching conditions.

(closes issue ASTERISK-19182)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350365 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The time passed by the LOCK function to an internal function was relative
time when the function expected absolute time.

* Don't use C++ keywords in get_lock().

(closes issue ASTERISK-16868)
Reported by: Andrey Solovyev
Patches:
      20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350273 65c4cc65-6c06-0410-ace0-fbb531ad65f3

There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3

The iax2_process_thread() can exit without anyone waiting to join the
thread.  If noone is waiting to join the thread then a large memory leak
occurs.

* Made iax2_process_thread() deatach itself if nobody is waiting to join
the thread.

(closes issue ASTERISK-17339)
Reported by: Tzafrir Cohen
Patches:
      asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch (license #5617) patch uploaded by Alex Villacis Lasso (modified)

(closes issue ASTERISK-17825)
Reported by: wangjin
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If a db_sync was scheduled just before shutdown, the atexit code calling
db_sync would have no effect, causing the astdb commit thread to stay
alive. This caused the SIP/realtime_sipregs test to fail. (The fallback
kill would run the atexit code again and that would wreak havoc.) This
fixes that the atexit kill condition is picked up properly.

(closes issue ASTERISK-18883)
Reviewed by: Terry Wilson

Review: https://reviewboard.asterisk.org/r/1659
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  r350127 | rmudgett | 2012-01-09 12:40:33 -0600 (Mon, 09 Jan 2012) | 12 lines
  
  Update contrib script live_ast to invoke Asterisk with valgrind and suppression file.
  
  * Added valgrind_compare script to compare two valgrind log files for
  differences.
  
  (issue ASTERISK-17339)
  Reported by: Tzafrir Cohen
  Patches:
        valgrind_compare (license #5035) script uploaded by Tzafrir Cohen
        live_ast_valgrind.diff (license #5035) patch uploaded by Tzafrir Cohen
        live_ast_valgrind_v2.diff (license #5185) patch uploaded by Paul Belanger
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  r350128 | rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 lines
  
  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/
  
  Merged revisions 326636 from http://svn.asterisk.org/svn/asterisk/branches/10
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The Asterisk -x command line parameter is documented inconsistently.

* Made the -x documentation and behavior consistent.

* Since this is also a new year, updated the copyright notices while here.

(closes issue ASTERISK-19094)
Reported by: Eugene
Patches:
      issueA19094_correct_asterisk_option_x.patch (license #5674) patch uploaded by Walter Doekes (modified)
Tested by: Eugene
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If SLA was reloaded without the config file being changed, current settings got
wiped out before the SLA reload code decided it wasn't going to reload the file
since nothing was changed.  Moving the settings reset later in the reload
process fixes this.

(closes issue AST-744)
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When someone does Set(CALLERPRES()=unavailable) (or
Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From header shows
"Anonymous" <anonymous@anonymous.invalid>. When sendrpid=yes/pai, the From
header will still display the callerid info, even though we supply an rpid
header with the anonymous info. It seems like we shouldn't leak that info in
any case. Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 seems
to indicate that one shouldn't send identifying info in the From in this case.

This patch anonymizes the From header as well even when sendrpid=yes/pai.

(closes issue ASTERISK-16538)

Review: https://reviewboard.asterisk.org/r/1649/
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A bug in the pbx_lua goto detection was causing the dialplan to hangup
unexpectedly after confbridge exited if it had called lua dialplan code during
execution.

Patch-by: Timo Teras
Acked-by: Matt Nicholson
(closes issue ASTERISK-18976)
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(closes issue ASTERISK-19055)
Reported by: Matt Jordan
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Even though we set the frame to the ast_null_frame and return that,
the caller of the frame hook may still need the frame.  This now is
a bit more careful about when it frees the frame, i.e., only under
the same conditions that applied when we duplicated it in the first
place.
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If a table is created by some other application and the primary key is not
named "AcctId", cel/cel_sqlite3_custom.c will always try to create the
table and fail because it already exists.

* Change the SQL table query to not require AcctId as the primary key.

(closes issue ASTERISK-18963)
Reported by: socketpair
Patches:
      fix.patch (license #6337) patch uploaded by socketpair
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Users created by users.conf with hasvoicemail=yes have been documented as
using a Gosub to stdexten since v1.6.0.  However, the code still generates
dialplan to access stdexten as a Macro as documented in v1.4; which does
not work with the newer extensions.conf.sample file.

* Make generated dialplan access the stdexten dialplan with the documented
Gosub instead of the older Macro style.

(closes issue ASTERISK-18809)
Reported by: Jay Allen
Patches:
      gosub_patch-pbx_config.patch (license #6323) patch uploaded by Jay Allen (modified)
Tested by: rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349782 65c4cc65-6c06-0410-ace0-fbb531ad65f3

ast_streamfile previously did unconditional seeking on files that broke
playback of formats that don't support that functionality.  This patch avoids
the seek that was causing the problem.  This regression was introduced in
r158062.

(closes issue ASTERISK-18994)
Patch-by: Timo Teras
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When receiving calls from a mobile phone into a DISA system on a connection with
significant interference, the reporter's Asterisk system would interpret DTMF incorrectly
and replicate digits received. This patch resolves that by increasing the number of
frames a mismatch has to be detected before assuming the DTMF is over by 1 frame and
adjusts dtmf_detect function to reset hits and misses only when an edge is detected.

(closes issue ASTERISK-17493)
Reported by: Alec Davis
Patches:
	bug18904-refactor.diff.txt uploaded by Alec Davis (license 5546)
Review: https://reviewboard.asterisk.org/r/1130/
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When catching a signal, in no fork mode the console thread is identical to the thread
responsible for catching the signal and closing Asterisk, which requires it to first
dispense with the console thread. Prior to this patch, if these threads were identical,
upon receiving a killing signal, the thread will send an URG signal to itself, which
we also catch and then promptly do nothing with. Obviously this isn't useful behavior.

(closes issue ASTERISK-19127)
Reported By: Bryon Clark
Patches:
	quit_on_signals.patch uploaded by Bryon Clark (license 6157)
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When looking up a ConfBridge profile, the config parser would, if it
found a channel datastore on the channel requesting the bridge profile,
unlock the channel mutex twice.  Since that's a little aggressive,
it now only unlocks it once.

(closes issue ASTERISK-19042)
Reported by: Matt Jordan
Tested by: Matt Jordan
Patches: 
  19042 uploaded by David Vossel (license 5628)
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A frame that is translated via ast_translate is also duplicated via ast_frdup.
This will allocate a new frame on the heap, which needs to be free'd
at the appropriate time.  This issue reporter used valgrind to find that this
occurred in res_fax's fax_gateway_framehook; a quick search through the code
showed that only place this was currently not handling the translatted frame
properly.

(closes issue ASTERISK-19133)
Reported by: Sylvain Rochet
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* Added NULL private pointer checks in the following chan_dahdi channel
callbacks: dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
dahdi_queryoption().

(closes issue ASTERISK-19142)
Reported by: Diego Aguirre
Tested by: rmudgett
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Previously, this init script would return 1 if Asterisk was already running.
This is incorrect behavior according to the LSB standard and has been fixed by
returning 0 instead.

(closes issue ASTERISK-17958)
Reported-by: johnc
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Added information collection from the output of the utilities: top, free, uptime, ifconfig
Added information collection from the output of the Asterisk command 'dahdi show status'
Added option / flag '-n, --non-interactive'
Updated man page to reflect new option / flag '-n, --non-interactive'

Patch-by: John Bigelow (itzanger)
(closes issue AST-749)
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(Closes issue ASTERISK-17953)
Reported by: George Konopacki
Patches:
	19400.patch uploaded by mmichelson (license 5049)
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(closes issue ASTERISK-18962)
reported by: Nir Simionovich
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Failing to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely.  This causes a variety of negative side
effects, depending on when the loop exits.  This patch handles the frame by
essentially swallowing the frame in the local loop, as the current channel
drivers expect the RTP bridge to handle the frame, and, in the case of the
local bridge loop, no additional action is necessary.

(issue ASTERISK-19040)
(issue ASTERISK-19128)
(issue ASTERISK-17725)
(issue ASTERISK-18340)
(closes issue ASTERISK-19095)
Reported by: Stefan Schmidt
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1640/
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of duplicating that logic.
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This commit removes the V.21 preamble detection code previously added to the
generic DSP implementation in Asterisk, and instead enhances the res_fax module
to be able to utilize V.21 preamble detection functionality made available by
FAX technology modules. This commit also adds such support to res_fax_spandsp,
which uses the Spandsp modem tone detection code to do the V.21 preamble
detection.

There should be no functional change here, other than much more reliable V.21
preamble detection (and thus T.38 gateway initiation).
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Merged revisions 349248 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349249 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Prior to this patch, res_musiconhold existed at the same module priority level
as the timing sources that it depends on.  This would cause a problem when
music on hold was reloaded, as the timing source could be changed after
res_musiconhold was processed.  This patch adds a new module priority level,
AST_MODPRI_TIMING, that the various timing modules are now loaded at.  This
now occurs before loading other resource modules, such that the timing source
is guaranteed to be set prior to resolving the timing source dependencies.

(closes issue ASTERISK-17474)
Reporter: Luke H
Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
Patches:
 asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026)
 asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026)
 asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026)

Review: https://reviewboard.asterisk.org/r/1578/
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Merged revisions 349194 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 349195 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349196 65c4cc65-6c06-0410-ace0-fbb531ad65f3

frames, even after all of the hooks are detached.  This patch short-cicuits us
out before we transcode unnecessarily.
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Merged revisions 349144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 349145 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349146 65c4cc65-6c06-0410-ace0-fbb531ad65f3

This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user.  It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.

(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1614/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3