[asterisk-commits] may: branch may/ooh323_qsig r397848 - in /team/may/ooh323_qsig: ./ apps/ apps...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Aug 27 15:41:14 CDT 2013
Author: may
Date: Tue Aug 27 15:41:03 2013
New Revision: 397848
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=397848
Log:
Multiple revisions 396201,396245,396253,396309,396311,396319-396321,396329,396331,396347,396365,396371,396378,396385,396391-396392,396401,396412,396417,396446,396462-396463,396474,396480,396490,396497-396498,396505,396512,396521,396528,396535,396542-396543,396552,396559-396560,396568
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r396201 | mjordan | 2013-08-05 23:01:45 +0400 (Mon, 05 Aug 2013) | 9 lines
Add AMI registration events for PJSIP outbound registration attempts
This patch adds AMI events whenever an outbound registration attempt succeeds
or fails from res_pjsip_outbound_registration. This brings it inline with
the existing SIP channel driver and IAX channel driver.
Review: https://reviewboard.asterisk.org/r/2729/
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r396245 | jrose | 2013-08-06 00:18:54 +0400 (Tue, 06 Aug 2013) | 13 lines
bridge features: Dial and Queue add features instead of replace them.
Dial and Queue would previously apply a new set of features whenever
bridging. These options would be based purely on the options supplied
to the dial/queue applications. This patch changes the function those
applications use to bridge calls so that the features will be added
to the set of existing features for each channel rather than having
them override the existing features.
(closes issue ASTERISK-22209)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2713/
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r396253 | elguero | 2013-08-06 00:20:41 +0400 (Tue, 06 Aug 2013) | 23 lines
Fix Registration Failure When A Peer And TLS Are Used
If a peer is used in a register line and TLS is defined as the transport, the
registration fails since the transport on the dialog is never set properly
resulting in UDP being used instead of TLS.
This patch sets the dialog's transport based on the transport that was defined
in the register line. If the register line does not specify a transport, the
parsing function for the register line always defaults back to UDP.
(closes issue ASTERISK-21964)
Reported by: Doug Bailey
Tested by: Doug Bailey
Patches:
asterisk-21964-set-reg-dialog-transport.diff
by Michael L. Young (license 5026)
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Merged revisions 396240 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 396248 from http://svn.asterisk.org/svn/asterisk/branches/11
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r396309 | wdoekes | 2013-08-06 12:36:15 +0400 (Tue, 06 Aug 2013) | 13 lines
Check result of ast_var_assign() calls for memory allocation failure.
We try to keep the system running even when all available memory is
spent.
Review: https://reviewboard.asterisk.org/r/2734/
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Merged revisions 396279 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 396287 from http://svn.asterisk.org/svn/asterisk/branches/11
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r396311 | wdoekes | 2013-08-06 12:43:22 +0400 (Tue, 06 Aug 2013) | 7 lines
Check result of ast_var_assign() calls for memory allocation failure (2).
Missed a spot in the previous commit.
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Merged revisions 396310 from http://svn.asterisk.org/svn/asterisk/branches/11
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r396319 | file | 2013-08-06 16:39:27 +0400 (Tue, 06 Aug 2013) | 9 lines
Fix crash in res_pjsip_outbound_registration when the remote server can not be resolved.
This crash was caused by decrementing the reference count of a newly created message when
it should not be. This change fixes that but also fixes all other cases where this was
incorrectly done.
(closes issue ASTERISK-22188)
Reported by: Kinsey Moore
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r396320 | kmoore | 2013-08-06 16:45:39 +0400 (Tue, 06 Aug 2013) | 8 lines
Fix memory leaks in the CDR engine
Fix refcount bugs and a possible locking problem in the CDR engine
relating to use of ao2_iterators.
Review: https://reviewboard.asterisk.org/r/2724/
(closes issue ASTERISK-22126)
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r396321 | kmoore | 2013-08-06 17:08:13 +0400 (Tue, 06 Aug 2013) | 8 lines
Expose res_pjsip threadpool options
Expose initial size, automatic increment, maximum size, and idle
timeout as configurable parameters for the res_pjsip thread pool.
Review: https://reviewboard.asterisk.org/r/2704/
(closes issue ASTERISK-22143)
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r396329 | dlee | 2013-08-06 18:28:23 +0400 (Tue, 06 Aug 2013) | 39 lines
Tweak caching topics to fix CEL tests
The Stasis changes in r395954 had an unanticipated side effect: messages
published directly to an _all topic does not get forwarded to the
corresponding caching topic.
This patch fixes that by changing how caching topics forward messages,
and how the caching pattern forwards are setup.
For the caching pattern, the all_topic is forwarded to the
all_topic_cached. This forwards messages published directly to the
all_topic to all_topic_cached.
In order to avoid duplicate messages on all_topic_cached, caching topics
were changed to no longer forward uncached messages. Subscribers to an
individual caching topic should only expect to receive cache updates,
and subscription change messages. Since individual caching topics are
new, this shouldn't be a problem.
There are a few minor changes to the pre-cache split behavior.
* For topics changed to use the caching pattern, the all_topic_cached
will forward snapshots in addition to cache updates. Since
subscribers by design ignore unexpected messages, this should be
fine.
* Caching topics that don't use the caching pattern no longer forward
non-cache updates. This makes no difference for the current caching
topics.
* mwi_topic_cached, channel_by_name_topic and
presence_state_topic_cached have no subscribers
* device_state_topic_cached's only subscriber only processes cache
udpates
(issue ASTERISK-22243)
Review: https://reviewboard.asterisk.org/r/2738
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r396331 | dlee | 2013-08-06 18:44:45 +0400 (Tue, 06 Aug 2013) | 23 lines
ARI: Add recording controls
This patch implements the controls from ARI recordings. The controls
are:
* DELETE /recordings/live/{recordingName} - stop recording and
discard it
* POST /recordings/live/{recordingName}/stop - stop recording
* POST /recordings/live/{recordingName}/pause - pause recording
* POST /recordings/live/{recordingName}/unpause - resume recording
* POST /recordings/live/{recordingName}/mute - mute recording (record
silence to the file)
* POST /recordings/live/{recordingName}/unmute - unmute recording.
Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.
(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/
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r396347 | dlee | 2013-08-07 01:20:58 +0400 (Wed, 07 Aug 2013) | 1 line
Fixed app_meetme for cache split changes
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r396365 | mjordan | 2013-08-08 01:38:17 +0400 (Thu, 08 Aug 2013) | 13 lines
Perform Ring-No-Answer checks before processing Hangup logic
The rna() routine will raise a Stasis message involving both the caller and the
agent. This doesn't work so well if we already hung up the agent channel, as
the channel doesn't quite exist. Not surprisingly, this will crash. This patch
properly runs the rna subroutine (performing all of the Ring-No-Answer logic)
prior to hanging up the agent channel.
(closes issue ASTERISK-22258)
Reported by: Kiril Valchev
Tested by: Kiril Valchev
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r396371 | mjordan | 2013-08-08 06:58:01 +0400 (Thu, 08 Aug 2013) | 7 lines
Handle Surrogate channels in Dial message processing
Depending on when a Surrogate channel replaces an existing channel, it is
possible to get a Dial message for the Surrogate channel. When this occurs, no
CDR will exist for the channel as Surrogate channels are ignored. Safely handle
the case when a CDR doesn't exist for a Dial message.
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r396378 | igorg | 2013-08-08 11:05:54 +0400 (Thu, 08 Aug 2013) | 7 lines
- Fix different issues with call transfer cancel. In case 3rd party busy or congestion call was not returned.
- Fix displaying soft button 'Redial' in case of no redial number exists
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Merged revisions 396377 from http://svn.asterisk.org/svn/asterisk/branches/11
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r396385 | kmoore | 2013-08-08 16:38:06 +0400 (Thu, 08 Aug 2013) | 10 lines
Prevent unreal channels from optimizing during DTMF emulation
This prevents unreal channel optimization during the prequalification
phase when either channel is involved in DTMF emulation. This prevents
a situation where an emulated digit would be missed because the
emulation was never completed.
Review: https://reviewboard.asterisk.org/r/2747/
(closes issue ASTERISK-22214)
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r396391 | mjordan | 2013-08-08 17:54:46 +0400 (Thu, 08 Aug 2013) | 10 lines
Prevent spurious memory error when appending backtrace with MALLOC_DEBUG
Backtraces are allocated outside of the usual memory tracking performed by
MALLOC_DEBUG. This allows them to be used by the memory tracking enabled
by that build option; however, it also means that when backtraces are
disposed of they have to be done so outside of the re-defined free.
This patch undef's free prior to disposing of the allocated backtrace when
a backtrace is appended as a result of 'core show locks'.
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r396392 | mjordan | 2013-08-08 18:13:05 +0400 (Thu, 08 Aug 2013) | 31 lines
Hide the Surrogate channels from external consumers; kill Masquerade events
This patch does three things:
1. It provides a Surrogate channel technology with a consolidated
"implementation detail flag" on the channel technology. This tells
consumers of Stasis that the creation of this channel is an implementation
detail in Asterisk and can be ignored (if they so choose). This
consolidates the conference recorder/announcer flags as well - these flags
had no additional meaning beyond "ignore this channel please".
2. It modifies allocation of a channel in two ways:
(a) If a channel technology can be determined from the name, we set it
directly in the allocation routine. This prevents the initial
publication of the message from going out with a NULL channel technology
where possible. This lets Stasis consumers get the right channel
technology on the first publication.
(b) It reorganizes allocation to make use of the 'finalized' property on the
channel. This was already used to know that a channel had completely
finished its construction in the masquerade routine; now we also use it
to know whether or not the setting of certain channel properties is
occurring during or post construction. The various set routines were
modified accordingly as well.
3. The masquerade event is now dead, Jim. It no longer served any purpose
whatsoever - if you perform a call pickup you'll get a Pickup event;
if you perform an attended transfer you will still get those events; if you
steal a channel to put it elsewhere you'll get the corresponding NewExten or
BridgeEnter events.
Review: https://reviewboard.asterisk.org/r/2740
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r396401 | rmudgett | 2013-08-08 21:51:26 +0400 (Thu, 08 Aug 2013) | 1 line
Remove some resolved or obsolete BUGBUG comments.
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r396412 | mjordan | 2013-08-08 22:40:15 +0400 (Thu, 08 Aug 2013) | 23 lines
Improve disk writes for wav49 format
Writing to a file in the wav49 format performs rather inefficiently. The
procedure is approximately:
(1) Write GSM frame to the end of the file
(2) Seek to the end of the file
(3) Seek to the header
(4) Update the file size
(5) Seek (again) to the end of the file
(6) Repeat
This pattern negates any attempt to use the stdio buffering setup in
ast_writefile. It also results in many small writes that require a seek going
to the disk each second which translates to poor disk performance on certain
file systems, particularly when there are multiple wav49 files being written
simultaneously.
(closes issue ASTERISK-19595)
Reported by: Byron Clark
Tested by: Byron Clark
patches:
gsm_wav_only_update_header_on_close.patch uploaded by byronclark (License 6157)
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r396417 | rmudgett | 2013-08-08 23:16:33 +0400 (Thu, 08 Aug 2013) | 10 lines
Make bridge snapshots use prefixes.
* Changed ast_manager_build_bridge_state_string() to assume an empty
prefix string just like ast_manager_build_channel_state_string().
* Created ast_manager_build_bridge_state_string_prefix() to work just like
ast_manager_build_channel_state_string_prefix().
* Made BridgeMerge AMI event use To/From prefixes.
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r396446 | wdoekes | 2013-08-09 00:23:26 +0400 (Fri, 09 Aug 2013) | 12 lines
Blocked revisions 396441
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Consistent memory allocation by ast_bt_get_symbols.
Always use ast_alloc/ast_free. This is handled differently in trunk (r391012).
Review: https://reviewboard.asterisk.org/r/2580/
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Merged revisions 396427 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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r396462 | rmudgett | 2013-08-09 00:51:38 +0400 (Fri, 09 Aug 2013) | 1 line
Remove extra CR/LF from AMI event.
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r396463 | rmudgett | 2013-08-09 00:52:49 +0400 (Fri, 09 Aug 2013) | 1 line
Add missing CR/LF to FakeMI stasis test AMI event.
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r396474 | tzafrir | 2013-08-09 02:09:07 +0400 (Fri, 09 Aug 2013) | 19 lines
chan_dahdi: create channels at run-time
This code adds chan_dahdi the command 'dahdi create channels <range>'
(where <range> is a single <n>-<m> or 'new') and updates 'dahdi destroy
channel' with a similar 'dahdi destroy channels'. It allows DAHDI
channels and spans to be added after the initial channel load
(without destroying all other channels as in 'dahdi restart').
It also includes some fixes to the D-Channel / span destruction code
(r394552).
This change is intended to provide a hook for a script running from
udev once a span has been assigned ("registered") / unassigned
("unregistered") for its channels. The udev hook configures the span's
channels with dahdi_cfg -S, and can then ask Asterisk to create ethe
channels. See the scripts added to DAHDI-tools in 2.7.0.
Review: https://reviewboard.asterisk.org/r/1598/
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r396480 | rmudgett | 2013-08-09 02:57:06 +0400 (Fri, 09 Aug 2013) | 1 line
Fix stasis/core unit test. Should have had the CR/LF.
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r396490 | mjordan | 2013-08-09 17:58:02 +0400 (Fri, 09 Aug 2013) | 5 lines
Update documentation for ConfBridge with some additional markup
Add some additional markup for items that needed it, e.g.,
replaceable tags, literal tags, etc.
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r396497 | jrose | 2013-08-09 21:22:28 +0400 (Fri, 09 Aug 2013) | 12 lines
bridge_channel: Support the lonely flag and make ARI use it.
The lonely flag is an optional flag for bridge channels that will
make them leave a bridge when a channel leaves if only lonely
channels are in the bridge at that point. This is useful for things
like ending recording and playback channels when they cease to be
interacting with other channels in the bridge.
(closes issue ASTERISK-22117)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2721/
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r396498 | jrose | 2013-08-09 21:28:15 +0400 (Fri, 09 Aug 2013) | 13 lines
pbx: Make originate threads indicate dial status when synchronous
This makes it so that we can detect failures to originate as with
earlier versions of Asterisk, which restores the Asterisk 11 behavior
for the originate manager action. This was causing the ACL tests for
SIP and IAX2 to fail since those tests expected originate failures
when ACLs would cause rejections. Also, this patch fixes crashes in
chan_sip when ACLs rejected peers during registration verification.
(closes issue ASTERISK-22212)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2753/
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r396505 | wdoekes | 2013-08-10 00:29:09 +0400 (Sat, 10 Aug 2013) | 4 lines
Don't leak frames when memory is full in autoservice_run.
Review: https://reviewboard.asterisk.org/r/2566/
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r396512 | rmudgett | 2013-08-10 01:50:08 +0400 (Sat, 10 Aug 2013) | 1 line
bridge_native_rtp: Remove some unnecessary NULL checks on c1.
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r396521 | mjordan | 2013-08-10 08:18:33 +0400 (Sat, 10 Aug 2013) | 5 lines
Unlock the dial operation lock on a failed dial
If a dial operation fails, the pbx_outgoing_attempt routine will exit without
first having unlocked the outgoing dial lock. This would be a "bad thing".
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r396528 | mjordan | 2013-08-10 23:45:31 +0400 (Sat, 10 Aug 2013) | 5 lines
Add some debugging when test_hashtab_thrash fails
Disabling DEBUG_THREADS caused this test to fail on the 32-bit build agent.
Adding some debugging to see why it thinks the test is timing out.
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r396535 | mjordan | 2013-08-11 00:29:56 +0400 (Sun, 11 Aug 2013) | 4 lines
Pipe test output through test object not stdout
Otherwise, it doesn't show up in the automated test failures
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r396542 | mjordan | 2013-08-12 19:48:58 +0400 (Mon, 12 Aug 2013) | 6 lines
Unlock outgoing dial lock on off nominal path
If the thread servicing the dial request isn't created successfully, the
outgoing dial lock will still be held when the function returns. This patch
unlocks the lock on this off nominal path.
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r396543 | mjordan | 2013-08-12 19:59:19 +0400 (Mon, 12 Aug 2013) | 42 lines
Fix two race conditions and ref counting issue when joining a bridge
These problems were all caught by a test in the Asterisk Test Suite that
originated some Local channels and attempted to move the ;2 half of the Local
channel into a bridge using the Bridge AMI action.
(1) When originating a channel, the Newchannel event is emitted quickly;
however, the ;2 channel will not have a pbx thread assigned to it until
after the outbound 'dialing' for the ;1 is complete. Thus, there is a period
of time where the outside world "knows" of the channel's existence and can
influence it but Asterisk has not yet started the dialplan execution thread.
If a Bridge AMI action is taken on the channel, the channel appears to be a
Dialed channel with no PBX thread; hence, the channel will be imparted into
the Bridge by first 'yanking' the channel. At the same time, a race condition
can occur after the yank (but before entering the bridge) when ;1 answers
and starts a PBX on the ;2. The end result currently is an assertion failure
in the Bridging API, as a channel with a PBX is imparted into the Bridge.
There's no way to prevent AMI from attempting to Bridge a channel
immediately after creation; likewise, holding the channel lock through the
entire Dial operation is unwise (and impossible). Instead of treating the
presence of a PBX thread as an error, we simply bail out of the adding the
channel to the bridge through ast_bridge_impart. The Bridge action will
then fail - but we avoid a situation where the channel is both executing
a PBX thread and simultaneously being given a separate thread in the
bridging system (which would be a "bad thing"). Since imparting a channel
with a PBX *can* occur and is not a programming error, the asserts have been
removed.
(2) When the first condition occurs, we have to take one of two actions: either
hangup the yanked channel as it did not enter the bridge, or deref it
because we don't own it. We can determine if we own it or not by testing
for the presence of the PBX thread. If we hung it up directly, we'd crash.
(3) bridge_find_channel does not increase the reference count of the
ast_bridge_channel object. The RAII_VAR usage in ast_bridge_add_channel
thus created a ticking time bomb in whatever bridge the channel moved into,
as the destructor for the ast_bridge_channel object would be called.
Review: https://reviewboard.asterisk.org/r/2741/
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r396552 | jbigelow | 2013-08-13 02:05:18 +0400 (Tue, 13 Aug 2013) | 13 lines
Add test suite events for when contacts are added or removed from an AOR
These are needed by the pjsip inbound registration test suite tests.
(issue ASTERISK-21833)
(issue ASTERISK-21834)
(issue ASTERISK-21835)
(issue ASTERISK-21837)
Review: https://reviewboard.asterisk.org/r/2700/
Review: https://reviewboard.asterisk.org/r/2739/
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r396559 | dlee | 2013-08-13 18:57:34 +0400 (Tue, 13 Aug 2013) | 8 lines
Fix build warnings when printf a tv_usec.
The debug logs added in r396528 neglected to account for suseconds_t
being an int.
See r392076 for more info.
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r396560 | dlee | 2013-08-13 19:11:44 +0400 (Tue, 13 Aug 2013) | 1 line
Missed a spot in r396559
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r396568 | dlee | 2013-08-13 19:27:32 +0400 (Tue, 13 Aug 2013) | 24 lines
ARI: allow other operations to happen while bridged
This patch changes ARI bridging to allow other channel operations to
happen while the channel is bridged.
ARI channel operations are designed to queue up and execute
sequentially. This meant, though, that while a channel was bridged,
any other channel operations would queue up and execute only after the
channel left the bridge.
This patch changes ARI bridging so that channel commands can execute
while the channel is bridged. For most operations, things simply work
as expected. The one thing that ended up being a bit odd is recording.
The current recording implementation will fail when one attempts to
record a channel that's in a bridge. Note that the bridge itself may
be recording; it's recording a specific channel in the bridge that
fails. While this is an annoying limitation, channel recording is
still very useful for use cases such as voice mail, and bridge
recording makes up much of the difference for other use cases.
(closes issue ASTERISK-22084)
Review: https://reviewboard.asterisk.org/r/2726/
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Merged revisions 396201,396245,396253,396309,396311,396319-396321,396329,396331,396347,396365,396371,396378,396385,396391-396392,396401,396412,396417,396446,396462-396463,396474,396480,396490,396497-396498,396505,396512,396521,396528,396535,396542-396543,396552,396559-396560,396568 from http://svn.asterisk.org/svn/asterisk/trunk
Removed:
team/may/ooh323_qsig/res/res_stasis_bridge_add.c
team/may/ooh323_qsig/res/res_stasis_bridge_add.exports.in
Modified:
team/may/ooh323_qsig/ (props changed)
team/may/ooh323_qsig/apps/app_meetme.c
team/may/ooh323_qsig/apps/app_minivm.c
team/may/ooh323_qsig/apps/app_playback.c
team/may/ooh323_qsig/apps/app_queue.c
team/may/ooh323_qsig/apps/app_stack.c
team/may/ooh323_qsig/apps/app_voicemail.c
team/may/ooh323_qsig/apps/confbridge/conf_chan_announce.c
team/may/ooh323_qsig/apps/confbridge/conf_chan_record.c
team/may/ooh323_qsig/apps/confbridge/conf_config_parser.c
team/may/ooh323_qsig/apps/confbridge/confbridge_manager.c
team/may/ooh323_qsig/bridges/bridge_native_rtp.c
team/may/ooh323_qsig/channels/chan_bridge_media.c
team/may/ooh323_qsig/channels/chan_dahdi.c
team/may/ooh323_qsig/channels/chan_sip.c
team/may/ooh323_qsig/channels/chan_unistim.c
team/may/ooh323_qsig/formats/format_wav_gsm.c
team/may/ooh323_qsig/funcs/func_frame_trace.c
team/may/ooh323_qsig/funcs/func_global.c
team/may/ooh323_qsig/funcs/func_strings.c
team/may/ooh323_qsig/include/asterisk/app.h
team/may/ooh323_qsig/include/asterisk/bridge.h
team/may/ooh323_qsig/include/asterisk/bridge_basic.h
team/may/ooh323_qsig/include/asterisk/bridge_features.h
team/may/ooh323_qsig/include/asterisk/bridge_internal.h
team/may/ooh323_qsig/include/asterisk/channel.h
team/may/ooh323_qsig/include/asterisk/core_unreal.h
team/may/ooh323_qsig/include/asterisk/frame.h
team/may/ooh323_qsig/include/asterisk/manager.h
team/may/ooh323_qsig/include/asterisk/stasis.h
team/may/ooh323_qsig/include/asterisk/stasis_app.h
team/may/ooh323_qsig/include/asterisk/stasis_app_impl.h
team/may/ooh323_qsig/include/asterisk/stasis_app_recording.h
team/may/ooh323_qsig/main/app.c
team/may/ooh323_qsig/main/autoservice.c
team/may/ooh323_qsig/main/bridge.c
team/may/ooh323_qsig/main/bridge_basic.c
team/may/ooh323_qsig/main/bridge_channel.c
team/may/ooh323_qsig/main/cdr.c
team/may/ooh323_qsig/main/cel.c
team/may/ooh323_qsig/main/channel.c
team/may/ooh323_qsig/main/channel_internal_api.c
team/may/ooh323_qsig/main/core_unreal.c
team/may/ooh323_qsig/main/features.c
team/may/ooh323_qsig/main/manager_bridges.c
team/may/ooh323_qsig/main/manager_channels.c
team/may/ooh323_qsig/main/pbx.c
team/may/ooh323_qsig/main/stasis_bridges.c
team/may/ooh323_qsig/main/stasis_cache.c
team/may/ooh323_qsig/main/stasis_cache_pattern.c
team/may/ooh323_qsig/main/utils.c
team/may/ooh323_qsig/pbx/pbx_dundi.c
team/may/ooh323_qsig/pbx/pbx_loopback.c
team/may/ooh323_qsig/res/ari/resource_bridges.c
team/may/ooh323_qsig/res/ari/resource_recordings.c
team/may/ooh323_qsig/res/ari/resource_recordings.h
team/may/ooh323_qsig/res/res_ari_recordings.c
team/may/ooh323_qsig/res/res_pjsip.c
team/may/ooh323_qsig/res/res_pjsip/config_system.c
team/may/ooh323_qsig/res/res_pjsip/include/res_pjsip_private.h
team/may/ooh323_qsig/res/res_pjsip_exten_state.c
team/may/ooh323_qsig/res/res_pjsip_messaging.c
team/may/ooh323_qsig/res/res_pjsip_notify.c
team/may/ooh323_qsig/res/res_pjsip_outbound_registration.c
team/may/ooh323_qsig/res/res_pjsip_registrar.c
team/may/ooh323_qsig/res/res_stasis.c
team/may/ooh323_qsig/res/res_stasis_playback.c
team/may/ooh323_qsig/res/res_stasis_recording.c
team/may/ooh323_qsig/res/stasis/control.c
team/may/ooh323_qsig/res/stasis/control.h
team/may/ooh323_qsig/rest-api/api-docs/recordings.json
team/may/ooh323_qsig/tests/test_hashtab_thrash.c
team/may/ooh323_qsig/tests/test_stasis.c
team/may/ooh323_qsig/utils/extconf.c
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- branch-11-blocked (original)
+++ branch-11-blocked Tue Aug 27 15:41:03 2013
@@ -1,1 +1,1 @@
-/branches/11:373240,375247,375702,385356,395020
+/branches/11:373240,375247,375702,385356,395020,396441
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.
Propchange: team/may/ooh323_qsig/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Aug 27 15:41:03 2013
@@ -1,1 +1,1 @@
-/trunk:1-380157,380165-391000,391012,391016-393400,393410-393530,393542-393834,393843-394050,394065-394089,394103-394600,394623-394881,394894-395298,395316-395900,395906-396200
+/trunk:1-380157,380165-391000,391012,391016-393400,393410-393530,393542-393834,393843-394050,394065-394089,394103-394600,394623-394881,394894-395298,395316-395900,395906-396568
Modified: team/may/ooh323_qsig/apps/app_meetme.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_meetme.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/app_meetme.c (original)
+++ team/may/ooh323_qsig/apps/app_meetme.c Tue Aug 27 15:41:03 2013
@@ -1167,7 +1167,7 @@
STASIS_MESSAGE_TYPE_INIT(meetme_talk_request_type);
meetme_event_message_router = stasis_message_router_create(
- ast_channel_cache());
+ ast_channel_topic_all_cached());
if (!meetme_event_message_router) {
meetme_stasis_cleanup();
Modified: team/may/ooh323_qsig/apps/app_minivm.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_minivm.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/app_minivm.c (original)
+++ team/may/ooh323_qsig/apps/app_minivm.c Tue Aug 27 15:41:03 2013
@@ -1674,7 +1674,7 @@
ast_channel_setoption(chan, AST_OPTION_RXGAIN, &record_gain, sizeof(record_gain), 0);
if (ast_test_flag(vmu, MVM_OPERATOR))
canceldtmf = "0";
- cmd = ast_play_and_record_full(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, global_silencethreshold, global_maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
+ cmd = ast_play_and_record_full(chan, playfile, recordfile, maxtime, fmt, duration, sound_duration, 0, global_silencethreshold, global_maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
if (record_gain)
ast_channel_setoption(chan, AST_OPTION_RXGAIN, &zero_gain, sizeof(zero_gain), 0);
if (cmd == -1) /* User has hung up, no options to give */
Modified: team/may/ooh323_qsig/apps/app_playback.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_playback.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/app_playback.c (original)
+++ team/may/ooh323_qsig/apps/app_playback.c Tue Aug 27 15:41:03 2013
@@ -220,6 +220,10 @@
s = x + 1;
ast_debug(2, "value is <%s>\n", s);
n = ast_var_assign("SAY", s);
+ if (!n) {
+ ast_log(LOG_ERROR, "Memory allocation error in do_say\n");
+ return -1;
+ }
AST_LIST_INSERT_HEAD(&head, n, entries);
/* scan the body, one piece at a time */
Modified: team/may/ooh323_qsig/apps/app_queue.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_queue.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/app_queue.c (original)
+++ team/may/ooh323_qsig/apps/app_queue.c Tue Aug 27 15:41:03 2013
@@ -4584,10 +4584,10 @@
case AST_CONTROL_BUSY:
ast_verb(3, "%s is busy\n", ochan_name);
ast_channel_publish_dial(qe->chan, o->chan, on, "BUSY");
- do_hang(o);
endtime = (long) time(NULL);
endtime -= starttime;
rna(endtime * 1000, qe, o->chan, on, membername, qe->parent->autopausebusy);
+ do_hang(o);
if (qe->parent->strategy != QUEUE_STRATEGY_RINGALL) {
if (qe->parent->timeoutrestart) {
start_time_tv = ast_tvnow();
Modified: team/may/ooh323_qsig/apps/app_stack.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_stack.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/app_stack.c (original)
+++ team/may/ooh323_qsig/apps/app_stack.c Tue Aug 27 15:41:03 2013
@@ -273,8 +273,9 @@
}
if (!found) {
- variables = ast_var_assign(var, "");
- AST_LIST_INSERT_HEAD(&frame->varshead, variables, entries);
+ if ((variables = ast_var_assign(var, ""))) {
+ AST_LIST_INSERT_HEAD(&frame->varshead, variables, entries);
+ }
pbx_builtin_pushvar_helper(chan, var, value);
} else {
pbx_builtin_setvar_helper(chan, var, value);
Modified: team/may/ooh323_qsig/apps/app_voicemail.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/app_voicemail.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/app_voicemail.c (original)
+++ team/may/ooh323_qsig/apps/app_voicemail.c Tue Aug 27 15:41:03 2013
@@ -14682,7 +14682,7 @@
ast_channel_setoption(chan, AST_OPTION_RXGAIN, &record_gain, sizeof(record_gain), 0);
if (ast_test_flag(vmu, VM_OPERATOR))
canceldtmf = "0";
- cmd = ast_play_and_record_full(chan, playfile, tempfile, maxtime, fmt, duration, sound_duration, silencethreshold, maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
+ cmd = ast_play_and_record_full(chan, playfile, tempfile, maxtime, fmt, duration, sound_duration, 0, silencethreshold, maxsilence, unlockdir, acceptdtmf, canceldtmf, 0, AST_RECORD_IF_EXISTS_OVERWRITE);
if (strchr(canceldtmf, cmd)) {
/* need this flag here to distinguish between pressing '0' during message recording or after */
canceleddtmf = 1;
Modified: team/may/ooh323_qsig/apps/confbridge/conf_chan_announce.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/confbridge/conf_chan_announce.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/confbridge/conf_chan_announce.c (original)
+++ team/may/ooh323_qsig/apps/confbridge/conf_chan_announce.c Tue Aug 27 15:41:03 2013
@@ -134,7 +134,7 @@
.send_text = ast_unreal_sendtext,
.queryoption = ast_unreal_queryoption,
.setoption = ast_unreal_setoption,
- .properties = AST_CHAN_TP_ANNOUNCER,
+ .properties = AST_CHAN_TP_INTERNAL,
};
struct ast_channel_tech *conf_announce_get_tech(void)
Modified: team/may/ooh323_qsig/apps/confbridge/conf_chan_record.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/confbridge/conf_chan_record.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/confbridge/conf_chan_record.c (original)
+++ team/may/ooh323_qsig/apps/confbridge/conf_chan_record.c Tue Aug 27 15:41:03 2013
@@ -86,7 +86,7 @@
.call = rec_call,
.read = rec_read,
.write = rec_write,
- .properties = AST_CHAN_TP_RECORDER,
+ .properties = AST_CHAN_TP_INTERNAL,
};
struct ast_channel_tech *conf_record_get_tech(void)
Modified: team/may/ooh323_qsig/apps/confbridge/conf_config_parser.c
URL: http://svnview.digium.com/svn/asterisk/team/may/ooh323_qsig/apps/confbridge/conf_config_parser.c?view=diff&rev=397848&r1=397847&r2=397848
==============================================================================
--- team/may/ooh323_qsig/apps/confbridge/conf_config_parser.c (original)
+++ team/may/ooh323_qsig/apps/confbridge/conf_config_parser.c Tue Aug 27 15:41:03 2013
@@ -123,8 +123,8 @@
<synopsis>Apply a denoise filter to the audio before mixing</synopsis>
<description><para>Sets whether or not a denoise filter should be applied
to the audio before mixing or not. Off by default. Requires
- codec_speex to be built and installed. Do not confuse this option
- with drop_silence. Denoise is useful if there is a lot of background
+ <literal>codec_speex</literal> to be built and installed. Do not confuse this option
+ with <replaceable>drop_silence</replaceable>. Denoise is useful if there is a lot of background
noise for a user as it attempts to remove the noise while preserving
the speech. This option does NOT remove silence from being mixed into
the conference and does come at the cost of a slight performance hit.
@@ -158,7 +158,7 @@
during mid sentence.
</para>
<para>
- 2. The drop_silence option depends on this value to
+ 2. The <replaceable>drop_silence</replaceable> option depends on this value to
determine when the user's audio should begin to be
dropped from the conference bridge after the user
stops talking. If this value is set too low the user's
@@ -200,7 +200,7 @@
room noise.
</para>
<para>
- 3. The drop_silence option depends on this value to determine
+ 3. The <replaceable>drop_silence</replaceable> option depends on this value to determine
when the user's audio should be mixed into the bridge
after periods of silence. If this value is too loose
the beginning of a user's speech will get cut off as they
@@ -274,15 +274,15 @@
Records the conference call starting when the first user
enters the room, and ending when the last user exits the room.
The default recorded filename is
- <filename>'confbridge-${name of conference bridge}-${start time}.wav</filename>
+ <filename>'confbridge-${name of conference bridge}-${start time}.wav'</filename>
and the default format is 8khz slinear. This file will be
- located in the configured monitoring directory in asterisk.conf.
+ located in the configured monitoring directory in <filename>asterisk.conf</filename>.
</para></description>
</configOption>
<configOption name="record_file" default="confbridge-${name of conference bridge}-${start time}.wav">
<synopsis>The filename of the conference recording</synopsis>
<description><para>
- When record_conference is set to yes, the specific name of the
+ When <replaceable>record_conference</replaceable> is set to yes, the specific name of the
record file can be set using this option. Note that since multiple
conferences may use the same bridge profile, this may cause issues
depending on the configuration. It is recommended to only use this
@@ -295,9 +295,9 @@
<configOption name="record_file_append" default="yes">
<synopsis>Append record file when starting/stopping on same conference recording</synopsis>
<description><para>
- When record_file_append is set to yes, stopping and starting recording on a
+ When <replaceable>record_file_append</replaceable> is set to yes, stopping and starting recording on a
conference adds the new portion to end of current record_file. When this is
- set to no, a new record_file is generated every time you start then stop recording
+ set to no, a new <replaceable>record_file</replaceable> is generated every time you start then stop recording
on a conference.
</para></description>
</configOption>
@@ -306,7 +306,7 @@
<description><para>
Sets how confbridge handles video distribution to the conference participants.
Note that participants wanting to view and be the source of a video feed
- _MUST_ be sharing the same video codec. Also, using video in conjunction with
+ <emphasis>MUST</emphasis> be sharing the same video codec. Also, using video in conjunction with
with the jitterbuffer currently results in the audio being slightly out of sync
with the video. This is a result of the jitterbuffer only working on the audio
stream. It is recommended to disable the jitterbuffer when video is used.</para>
@@ -395,7 +395,7 @@
<configObject name="menu">
<synopsis>A conference user menu</synopsis>
<description>
- <para>Conference users, as defined by a <literal>conf_user</literal>,
+ <para>Conference users, as defined by a <replaceable>conf_user</replaceable>,
can have a DTMF menu assigned to their profile when they enter the
<literal>ConfBridge</literal> application.</para>
</description>
@@ -412,7 +412,7 @@
</configOption>
<configOption name="^[0-9A-D*#]+$">
<synopsis>DTMF sequences to assign various confbridge actions to</synopsis>
- <description><para>--- ConfBridge Menu Options ---</para>
+ <description>
<para>The ConfBridge application also has the ability to apply custom DTMF menus to
[... 4526 lines stripped ...]
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