Starting Asterisk would kick back an ERROR message stating that the Stasis
message type ast_channel_snapshot_type was used prior to initialization.
This occurred due to the caching topic being created prior to the message
type that it depended on.
This patch re-orders the start up such that the message type is initialized
prior to the caching topic. It also checks the return value of the
initialization of the agent login/logoff types.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
DTMF start/end and hold/unhold events have state because a DTMF begin
event and hold event must be ended by something.
The following cases need to be handled when a channel is moved around in
the system.
* When a channel leaves a bridge it may owe a DTMF end event to the
bridge.
* When a channel leaves a bridge it may owe an UNHOLD event to the bridge.
(This case is explicitly ignored because things like transfers need
explicit control over this.)
* When a channel leaves the bridging system it may need to simulate a DTMF
end event to the channel.
* When a channel leaves the bridging system it may need to simulate an
UNHOLD event to the channel.
The patch also fixes the following:
* Fixes playing a file and restarting MOH using the latest MOH class used.
(closes issue ASTERISK-22043)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2791/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Review https://reviewboard.asterisk.org/r/2580/ tried to fix the mismatch
in memory pools but had a math error determining the buffer size and
didn't address other similar memory pool mismatches.
* Effectively reverted the previous patch to go in the same direction as
trunk for the returned memory pool of ast_bt_get_symbols().
* Fixed memory leak in ast_bt_get_symbols() when BETTER_BACKTRACES is
defined.
* Fixed some formatting in ast_bt_get_symbols().
* Fixed sig_pri.c freeing memory allocated by libpri when MALLOC_DEBUG is
enabled.
* Fixed __dump_backtrace() freeing memory from ast_bt_get_symbols() when
MALLOC_DEBUG is enabled.
* Moved __dump_backtrace() because of compile issues with the utils
directory.
(closes issue ASTERISK-22221)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2778/
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Merged revisions 397525 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 397528 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If an option is registered to a type and it is the last known type in the list
of registered types, and the option fails to register, an overrun of the types
array can occur due to the index variable having been already incremented.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds pass through support for Opus and VP8. That includes:
* Format attribute negotiation for Opus. Note that unlike some other codecs,
the draft RFC specifies having spaces delimiting the attributes in addition
to ';', so you have "attra=X; attrb=Y". This broke the attribute parsing in
chan_sip, so a small tweak was also included in this patch for that.
* A format attribute negotiation module for Opus, res_format_attr_opus
* Fast picture update for VP8. Since VP8 uses a different RTCP packet number
than FIR, this really is specific to VP8 at this time.
Note that the format attribute negotiation in res_pjsip_sdp_rtp was written
by mjordan. The rest of this patch was written completely by Lorenzo Miniero.
Review: https://reviewboard.asterisk.org/r/2723/
(closes issue ASTERISK-21981)
Reported by: Tzafrir Cohen
patches:
asterisk_opus+vp8_passthrough_20130718.patch uploaded by lminiero (License 6518)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397526 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are times when a configuration option should not have documentation.
1. Some options are registered with a particular object merely as a warning to
users. These options aren't even really 'deprecated' - which has its own
separate API call - they are actually provided by a different configuration
file. The options are merely registered so that the user gets a warning that
a different configuration file provides the item.
2. Some object types - most notably some used by modules that use sorcery - are
completely internal and should never be shown to the user.
3. Sorcery itself has several 'hidden' fields that should never be shown to a
user.
This patch updates the configuration framework and sorcery with additional API
calls that allow a module to register types as internal and options as not
requiring documentation. This bypasses the XML documentation checking.
This patch also re-enables the strict XML documentation checking in trunk, as
well as updates some documentation that was missing.
Review: https://reviewboard.asterisk.org/r/2785/
(closes issue ASTERISK-22359)
Reported by: Matt Jordan
(closes issue ASTERISK-22112)
Reported by: Rusty Newton
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.
Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The cause code needs to be passed from the disconnecting channel to the
bridge peers if the disconnecting channel dissolves the bridge.
* Made the call to an app_agent_pool agent disconnect with the busy cause
code if the agent does not ack the call in time or hangs up before acking
the call.
(closes issue ASTERISK-22042)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2772/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This essentially makes app_queue usable again. From reviewboard:
* Reporting of transfers and call completion is done by creating stasis
subscriptions and listening for specific events in order to determine
when the call is finished (either via a transfer or hangup).
* Dial end messages have been added where they were previously missing.
* Queue stats are properly being updated again once calls have finished.
* AgentComplete stasis messages and AMI events are now occurring again.
* Mixmonitor starting has been factored into its own function and uses the
Mixmonitor API now instead of using ast_pbx_run()
In addition to the changes in app_queue, there are several supplementary changes as well:
* Queue logging now differentiates between attended and blind transfers. A
note about this is in the CHANGES file.
* Local channel optimization events now report more information. This
includes which of the two local channels involved is the destination of
the optimization, the channel that is replacing the destination local channel,
and an identifier so that begin and end events can be matched to each other.
The end events are now sent whether the optimization was successful or not and
includes an indicator of whether the optimization was successful.
* Changes were made to features and bridging_basic so that additional flags may
be set on a bridge. This is necessary because the queue requires that its
bridge only allows move-swap local channel optimizations into the bridge.
(closes issue ASTERISK-21517)
Reported by Matt Jordan
(closes issue ASTERISK-21943)
Reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2694
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Resync the abstract jitter buffer on the following additional control
frames:
AST_CONTROL_HOLD
AST_CONTROL_UNHOLD
AST_CONTROL_T38_PARAMETERS
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This modifies the behavior of the CEL engine to conform to documented
behavior for Asterisk 12 as defined on the wiki
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CEL+Specification
The primary changes deal with removal of the peer field from function
calls since it is no longer directly relevant to the bridging system
and removal of the layer of CDR-like business logic that was providing
a partial emulation of Asterisk 11 CEL functionality. With this change,
there is no longer a distinction between "bridges" and "conferences"
and all participation changes are denoted with bridge enter and bridge
exit messages.
This updates the CEL unit tests to handle these changes and simplifies
some of the macros used in the process.
This also fixes a segfault when attempting to ref a configuration that
failed to load.
Review: https://reviewboard.asterisk.org/r/2788/
(issue ASTERISK-21567)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The --version-script,asterisk.exports linker flag (and the module
exports) didn't provide _IO_stdin_used in the list of exported symbols.
That causes some kind of libc compatibility mode to kick in, where
stdio file structures (stdout/stderr) land somewhere else. In the
case of the Sparc, they landed on misaligned memory.
This became apparent first after r376428 (Reorder startup sequence)
when a lot of ast_log's were replaced with fprintf's. Writing to
stderr triggered a SIGBUS. (Compared to x86 and amd64 architectures,
the Sparc is very picky about memory alignment.)
(issue ASTERISK-21763)
(issue ASTERISK-21665)
Reported by: Jeremy Kister
Review: https://reviewboard.asterisk.org/r/2760/
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Merged revisions 397377 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 397378 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the file udptl.conf is unavailable at startup, UDPTL will fail to
initialize and while it makes some noise, it isn't immediately
obvious why consumers start to fail when using it. This patch makes
UDPTL load as though an empty config was provided when udptl is
unavailable at startup.
(closes issue ASTERISK-22349)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2773/
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Merged revisions 397365 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For times when a reference in ARI might be ambiguous, the reference is
built as an URI (such as channel:1376341790.3).
An endpoint's channel list is not ambiguous, and in fact the field is
named 'channel_ids', but it had channel URI's instead of channel id's.
This patch changes the list to be the raw id instead of the URI.
(closes issue ASTERISK-22291)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added an option flags parameter to interval hooks. Interval hooks now
can specify if the callback will affect the media path or not.
* Added an option flags parameter to the bridge action custom callback.
The action callback now can specify if the callback will affect the media
path or not.
* Made the holding bridge technology reexamine the participant idle mode
option whenever the entertainment is restarted.
* Fixed app_agent_pool waiting agents needlessly starting and stopping MOH
every second by specifying the heartbeat interval hook as not affecting
the media path.
* Fixed app_agent_pool agent alert from restarting the MOH after the alert
beep. The agent entertainment is now changed from MOH to silence after
the alert beep.
* Fixed holding bridge technology to defer starting the entertainment. It
was previously a mixture of immediate and deferred.
* Fixed holding bridge technology to immediately stop the entertainment.
It was previously a mixture of immediate and deferred. If the channel
left the bridging system, any deferred stopping was discarded before
taking effect.
* Miscellaneous holding bridge technology rework coding improvements.
Review: https://reviewboard.asterisk.org/r/2761/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Performing a blond transfer (attended transfer that is completed
before the transfer recipient picks up) externally through chan_sip
or chan_pjsip would result in lost references to the channels
involved with the transfer as well as their bridge.
(closes issue ASTERISK-22092)
Reported by: mmichelson
Review: https://reviewboard.asterisk.org/r/2766/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It is not safe to iterate over a macro'd list of ao2 objects, deref them such
that the item's destructor is called, and leave them in the list. The list
macro to iterate over items requires the item to be a valid allocated object
in order to proceed to the next item; with MALLOC_DEBUG on the corruption of
the linked list is caught in the crash.
This patch fixes the invalid access to free'd memory by removing the ao2 item
from the list before de-refing it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.
This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.
Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.
Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP/foo -- Local;1==Local;2 -- .... -- Local;1==Local;2 -- SIP/bar
Kick a ;1 channel and the chain toward SIP/foo goes away.
Kick a ;2 channel and the chain toward SIP/bar goes away.
This can leave a local channel chain between the kicked ;1 and ;2 channels
that are orphaned until you manually request one of those channels to
hangup or request the bridge to dissolve.
* Added ast_bridge_kick() as a companion to ast_bridge_remove(). The
functional difference is that ast_bridge_kick() may dissolve the bridge as
a result of the channel leaving the bridge.
* Made CLI "bridge kick <bridge> <channel>" use ast_bridge_kick() instead
of ast_bridge_remove() so the bridge can dissolve if needed.
* Renamed bridge_channel_handle_hangup() to ast_bridge_channel_kick() and
made it accessible to other files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The horrid structure of the source in the utils directory strikes again.
Moved the _ast_mem_backtrace_buffer[] definition from the logical location
in utils.c to hashtab.c so the aelparse and conf2ael utilities can link.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change r395954 reordered some stasis object destruction, which should
have been fine. Unfortunately, it caused some hard to reproduce issues
related to objects being accessed after they had been destroyed. The
patch in r396329 fixed the destruction order problem; this patch
addresses the underlying issue. A few other stasis-related fixes were
also added.
* Add ref-bumps around areas where objects may get transitively
destroyed. (For example, where we lock a topic, unref a subscription,
which unrefs the topic, which explodes the topic when we try to
unlock it.)
* Wrote an extensive doxygen page about Stasis implementation,
relationships between objects, lifecycles of objects, how the
refcounting works, etc. Many other comments were added, corrected, or
cleaned up.
* Added an assert to the topic dtor to catch extra ref decrements.
* Fixed type used after destruction errors for graceful shutdown in
stasis_channels.c.
* I added two unit tests in an attempt to catch destruction order
issues. Since the underlying cause is a race condition, though, the
tests rarely failed even when the code was wrong.
* Fixed a leak in stasis_cache_pattern.c.
(closes issue ASTERISK-22243)
Review: https://reviewboard.asterisk.org/r/2746/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This reworks the CLI commands used to access sounds information from
"sounds show[ soundid]" to "core show sounds" and
"core show sound <soundid>". This also reworks the "sounds reload" CLI
command to fall under normal module reloading ("module reload sounds").
Also, make trunk build when DEBUG_MALLOC is not enabled.
Review: https://reviewboard.asterisk.org/r/2745/
(closes issue ASTERISK-22141)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When asterisk has run out of memory (for whatever reason), the alloc
function logs a message. Logging requires memory. A recipe for
infinite recursion.
Stop the recursion by comparing the function call depth for sane values
before attempting another OOM log message.
Review: https://reviewboard.asterisk.org/r/2743/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
By their nature, the connected line and redirecting interception routines
are not supposed to affect the channel's media. Therefore, they should
not suspend and unsuspend the channel while running. The
suspend/unsuspend operations could be expensive depending upon the bridge
and channel technology involved.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396814 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Bridge API DTMF hook matching would not deal with DTMF end events
only. It required a DTMF begin event to start matching the DTMF hooks.
There are many places in Asterisk where code only generates DTMF end
events without the corresponding begin event. One such place is the AMI
action Atxfer.
* Fixed DTMF hook matching if there is a string of DTMF frames in the read
queue. We could potentially miss some of them before.
* Fixed AMI Atxfer action documentation.
(closes issue ASTERISK-22037)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2752/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396732 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The feature_attended_transfer test is failing due to Asterisk not
passing DTMF in the bridges created for internal attended transfers.
This sets the features initialization routine to set this flag by
default and adjusts the basic bridge and confbridge's use of the
bridging system accordingly as per Richard's suggestion instead of
adjusting this individual case. This change allows the necessary DTMF
to pass through the attended transfer bridge and complete the test
successfully.
Review: https://reviewboard.asterisk.org/r/2759/
(closes issue ASTERISK-22222)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Mark Michelson
Matthew Jordan
Richard Mudgett
Kinsey Moore
Walter Doekes
Jonathan Rose
David M. Lee