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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* 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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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'.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Missed a spot in the previous commit.
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Merged revisions 396310 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(related to issue ASTERISK-21903)
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Merged revisions 396199 from http://svn.asterisk.org/svn/asterisk/branches/11
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indicate when streaming an audio file fails like it is done in other parts
of the code to indicate an error.
Note was requested by Paul Belanger:
http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html
(related to issue ASTERISK-21903)
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Merged revisions 396196 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 396197 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396198 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Suspend and unsuspend callbacks are added to the holding bridge so
that entertainment can be disabled and re-enabled when operations
would suspend a channel on the bridge (such as playback operations).
This fixes entertainment so that when those operations end, the
entertainment can pick back up and it also serves as an optimization.
Also, this patch fixes a bug caused by triggering ringing frames
immediately instead of pushing them to the queue which created a race
condition where sometimes parking with ringing during attended
transfers would cause the ringing to be interrupted by an unhold
frame.
(closes issue ASTERISK-22006)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2711/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396189 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Roles are now cleared with each entry into a bridge with addChannel.
If the roles parameter is present, the role specified will be applied
to all channels being added with the addChannel command.
(closes issue ASTERISK-21973)
Reported by: Matt Jordan
https://reviewboard.asterisk.org/r/2691/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Adds the following unit tests:
* create_lot: tests adding and removal of a new parking lot (baseline)
* park_extensions: creates a parking lot that registers extensions and
then confirms that all of the expected extensions exist
* extensions_conflicts: creates numerous parking lots to test that
extension conflicts in parking lots result in parking lot
creation failing
* dynamic_parking_variables: Tests that the creation of dynamic
parking lots respects the related channel variables set on the
channel that requests them.
* park_call: Tests adding a channel to a parking lot's holding bridge
by standard parking functions.
* retrieve_call: Tests pulling a channel out of a parking lot's
holding bridge via parked call retrieval functions.
(closes issue ASTERISK-22138)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2714/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.
This patch renames the variables, adding the ast_ prefix so they will
be exported.
Review: https://reviewboard.asterisk.org/r/2737
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The AMI message router is owned wholly by manager.c. Previously, each of the
manager_{item} source files had their own message router and they unsubscribed
from each; once they moved over to using a single message router only a single
unsubscribe became necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds basic system information access to ARI.
The results are roughly what you get from 'core show settings', with a
few minor differences.
* Data is structured, with 'build', 'system', 'config' and 'status'
sub-objects.
* Each sub-object is selectable, using the ?only= parameter. A comma
separated list can be provided to select multiple sections.
* A few config options are numeric, for which 0 means 'unlimited'.
Instead of having a special interpretation of those fields, they
are simply omitted if they're 0.
* The information is limited to what might be useful to building
external applications.
(closes issue ASTERISK-21575)
Review: https://reviewboard.asterisk.org/r/2702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Swagger allows parameters to be specified as 'allowMultiple', meaning
that the parameter may be specified as a comma separated list of
values.
I had written some of the API docs using that, but promptly forgot
about implementing it. This patch finally fills in that gap.
The codegen template was updated to represent 'allowMultiple' fields
as array/size fields in the _args structs. It also parses the comma
separated list using ast_app_separate_args(), so quoted strings in the
argument will be handled properly.
Review: https://reviewboard.asterisk.org/r/2698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In tracking down some unit tests failures, I ended up reading the fine
print[1] regarding Jansson's thread safety.
In short:
1. Ref-counting is non-atomic.
2. json_dumps() and friends are not thread safe.
This patch adds locking where necessary to our ast_json_* wrapper API,
with documentation in json.h describing the thread safety limitations of
the API.
[1]: http://www.digip.org/jansson/doc/2.4/portability.html#thread-safety
Review: https://reviewboard.asterisk.org/r/2716/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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occurring.
* BridgeEnter now contains the unique ID of the channel that is to be swapped out, if applicable.
* There is a ParkedCallSwap event that is sent when a parked channel has a new channel take its place.
(closes issue ASTERISK-22193)
reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/2712
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This moves ast_str_container_alloc, ast_str_container_add,
ast_str_container_remove, and related private functions into
strings.c/h since they really don't belong in astobj2.c/h.
As a result of this move, utils also had to be updated.
Review: https://reviewboard.asterisk.org/r/2719/
(closes issue ASTERISK-22041)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This commit is smaller than the initial review placed on review board. This is because
a change to allow for channel drivers to access parking functionality externally was
committed and invalidated quite a few of the changes initially made.
(closes issue ASTERISK-22039)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2717
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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For chan_pjsip, this introduces CLI/AMI remote unregistration commands,
reworks CLI syntax for sending NOTIFYs, adds AMI qualification support,
and adds documentation for PJSIPNotify.
This also fixes two refcounting bugs in the outbound registration code.
Review: https://reviewboard.asterisk.org/r/2695/
(closes issue ASTERISK-21939)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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More missing include files. :-\
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch does the following:
* It moves the pickup code out of features.c and into pickup.c
* It removes the vast majority of dead code out of features.c. In particular,
this includes the parking code.
(issue ASTERISK-22134)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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contains '*'.
(closes issue AST-1198)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Some regex implementations won't compile an empty string. Assuming that
it's equivalent of a regex that will match anything, use ".?" instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch does the following:
* It adds support for externally initiated parking requests. In particular,
chan_skinny has a protocol level message that initiates a call park.
This patch now supports that option, as well as the protocol specific
mechanisms in chan_dahdi/sig_analog and chan_mgcp.
* A parking bridge features virtual table has been added that provides
access to the parking functionality that the Bridging API needs. This
includes requests to park an entire 'call' (with little or no additional
information, thank you chan_skinny), perform a blind transfer to a parking
extension, determine if an extension is a parking extension, as well as the
actual "do the parking" request from the Bridging API.
* Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
functions
* The removal of some - but not all - dead parking code from features.c
This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)
Review: https://reviewboard.asterisk.org/r/2710
(closes issue ASTERISK-22134)
Reported by: Matt Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch adds the ability in Queue to raise a hint when a member's paused
state changes. The hint uses the form 'Queue:{queue_name}_pause_{member_name}',
where {queue_name} and {member_name} are the name of the queue and the name
of the member to subscribe to, respectively.
For example: exten => 8501,hint,Queue:sales_pause_mark.
Members will show as In Use when paused.
Note that the format of the queue pause hint was changed slightly from what
is on the issue to accomodate suggestion on the code review.
Review: https://reviewboard.asterisk.org/r/2254
(closes issue ASTERISK-20842)
Reported by: Philippe Lindheimer
patches:
qpause-10-378206.diff uploaded by Philippe Lindheimer (license 5519)
qpause-11-378206.diff uploaded by Philippe Lindheimer (license 5519)
qpause-trunk-378206.diff uploaded by Philippe Lindheimer (license 5519)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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