Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
These are fixes for compilation under gcc 5.0...
chan_sip.c: In parse_request needed to make 'lim' unsigned.
inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99
inline semantics (same as clang).
ccss.c: In ast_cc_set_parm, needed to fix weird comparison.
dsp.c: Needed to work around a possible compiler bug. It was throwing
an array-bounds error but neither
sgriepentrog, rmudgett nor I could figure out why.
manager.c: In action_atxfer, needed to correct an array allocation.
This patch will go to 11, 13, trunk.
Review: https://reviewboard.asterisk.org/r/4581/
Reported-by: Jeffrey Ollie
Tested-by: George Joseph
ASTERISK-24932 #close
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Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups. Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe. ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.
Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.
ASTERISK-24142 #close
Reported by: David Brillert
ASTERISK-24683 #close
Reported by: Peter Katzmann
ASTERISK-24805 #close
Reported by: Badalian Vyacheslav
ASTERISK-24881 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
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This introduces stasis.conf and a mechanism to prevent certain message
types from being published. Internally, this works by preventing the
chosen message types from being created which ensures that those
message types can never be published. This patch also adjusts message
publishers such that message payloads are not created if the related
message type is not available.
ASTERISK-23943 #close
Review: https://reviewboard.asterisk.org/r/3823/
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Much needed was a way to assign id to objects on creation, and
much change was necessary to accomplish it. Channel uniqueids
and linkedids are split into separate string and creation time
components without breaking linkedid propgation. This allowed
the uniqueid to be specified by the user interface - and those
values are now carried through to channel creation, adding the
assignedids value to every function in the chain including the
channel drivers. For local channels, the second channel can be
specified or left to default to a ;2 suffix of first. In ARI,
bridge, playback, and snoop objects can also be created with a
specified uniqueid.
Along the way, the args order to allocating channels was fixed
in chan_mgcp and chan_gtalk, and linkedid is no longer lost as
masquerade occurs.
(closes issue ASTERISK-23120)
Review: https://reviewboard.asterisk.org/r/3191/
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Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.
Review: https://reviewboard.asterisk.org/r/2879
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r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
Minor performance bump by not allocate manager variable struct if we don't need it
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r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
Stasis performance improvements
This patch addresses several performance problems that were found in
the initial performance testing of Asterisk 12.
The Stasis dispatch object was allocated as an AO2 object, even though
it has a very confined lifecycle. This was replaced with a straight
ast_malloc().
The Stasis message router was spending an inordinate amount of time
searching hash tables. In this case, most of our routers had 6 or
fewer routes in them to begin with. This was replaced with an array
that's searched linearly for the route.
We more heavily rely on AO2 objects in Asterisk 12, and the memset()
in ao2_ref() actually became noticeable on the profile. This was
#ifdef'ed to only run when AO2_DEBUG was enabled.
After being misled by an erroneous comment in taskprocessor.c during
profiling, the wrong comment was removed.
Review: https://reviewboard.asterisk.org/r/2873/
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r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
Taskprocessor optimization; switch Stasis to use taskprocessors
This patch optimizes taskprocessor to use a semaphore for signaling,
which the OS can do a better job at managing contention and waiting
that we can with a mutex and condition.
The taskprocessor execution was also slightly optimized to reduce the
number of locks taken.
The only observable difference in the taskprocessor implementation is
that when the final reference to the taskprocessor goes away, it will
execute all tasks to completion instead of discarding the unexecuted
tasks.
For systems where unnamed semaphores are not supported, a really
simple semaphore implementation is provided. (Which gives identical
performance as the original taskprocessor implementation).
The way we ended up implementing Stasis caused the threadpool to be a
burden instead of a boost to performance. This was switched to just
use taskprocessors directly for subscriptions.
Review: https://reviewboard.asterisk.org/r/2881/
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r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
Optimize how Stasis forwards are dispatched
This patch optimizes how forwards are dispatched in Stasis.
Originally, forwards were dispatched as subscriptions that are invoked
on the publishing thread. This did not account for the vast number of
forwards we would end up having in the system, and the amount of work it
would take to walk though the forward subscriptions.
This patch modifies Stasis so that rather than walking the tree of
forwards on every dispatch, when forwards and subscriptions are changed,
the subscriber list for every topic in the tree is changed.
This has a couple of benefits. First, this reduces the workload of
dispatching messages. It also reduces contention when dispatching to
different topics that happen to forward to the same aggregation topic
(as happens with all of the channel, bridge and endpoint topics).
Since forwards are no longer subscriptions, the bulk of this patch is
simply changing stasis_subscription objects to stasis_forward objects
(which, admittedly, I should have done in the first place.)
Since this required me to yet again put in a growing array, I finally
abstracted that out into a set of ast_vector macros in
asterisk/vector.h.
Review: https://reviewboard.asterisk.org/r/2883/
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r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
Remove dispatch object allocation from Stasis publishing
While looking for areas for performance improvement, I realized that an
unused feature in Stasis was negatively impacting performance.
When a message is sent to a subscriber, a dispatch object is allocated
for the dispatch, containing the topic the message was published to, the
subscriber the message is being sent to, and the message itself.
The topic is actually unused by any subscriber in Asterisk today. And
the subscriber is associated with the taskprocessor the message is being
dispatched to.
First, this patch removes the unused topic parameter from Stasis
subscription callbacks.
Second, this patch introduces the concept of taskprocessor local data,
data that may be set on a taskprocessor and provided along with the data
pointer when a task is pushed using the ast_taskprocessor_push_local()
call. This allows the task to have both data specific to that
taskprocessor, in addition to data specific to that invocation.
With those two changes, the dispatch object can be removed completely,
and the message is simply refcounted and sent directly to the
taskprocessor.
Review: https://reviewboard.asterisk.org/r/2884/
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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)
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In the move from Asterisk's event system to Stasis, this makes
distributed device state aggregation always-on, removes unnecessary
task processors where possible, and collapses aggregate and
non-aggregate states into a single cache for ease of retrieval. This
also removes an intermediary step in device state aggregation.
Review: https://reviewboard.asterisk.org/r/2389/
(closes issue ASTERISK-21101)
Patch-by: Kinsey Moore <kmoore@digium.com>
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Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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Update and extend the configuration_file group and enable linking. Commit other cleanups from multi-version Doxygen testing. Update title that was left behind many years ago.
(issue ASTERISK-20259)
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Passing an ast_str pointer by value that then calls
ast_str_set(), ast_str_set_va(), ast_str_append(), or
ast_str_append_va() can result in the pointer originally
passed by value being invalidated if the ast_str had
to be reallocated.
This fixes places in the code that do this. Only the
example in ccss.c could result in pointer invalidation
though since the other cases use a stack-allocated ast_str
and cannot be reallocated.
I've also updated the doxygen in strings.h to include
notes about potential misuse of the functions mentioned
previously.
Review: https://reviewboard.asterisk.org/r/2161
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For each item in core_instances disposed of in the shutdown of ccss, any
generic monitor instances referenced by the objects will be removed from
generic_monitors during their destruction. Hilarity ensues if
generic_monitors no longer exists.
Thanks to the Asterisk Test Suite's generic_ccss test for complaining loudly
when it ran into this.
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* Whitespace, doc-blocks, spelling, case, missing and incorrect tags.
* Add cleanup to Makefile for the Doxygen configuration update
* Start updating Doxygen configuration for cleaner output
* Enable inclusion of configuration files into documentation
* remove mantisworkflow...
* update documentation README
* Add markup to Tilghman's email and talk with him about updating his email, he knows...
* no code changes on this commit other than the mentioned Makefile change
(issue ASTERISK-20259)
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Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style. Some missing txt file links are removed but their content or essense will be included in some later updates. A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen.
Further updates coming.
(issue ASTERISK-20259)
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r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
Add support-level indications to many more source files.
Since we now have tools that scan through the source tree looking for files
with specific support levels, we need to ensure that every file that is
a component of a 'core' or 'extended' module (or the main Asterisk binary)
is explicitly marked with its support level. This patch adds support-level
indications to many more source files in tree, but avoids adding them to
third-party libraries that are included in the tree and to source files
that don't end up involved in Asterisk itself.
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r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
Add a script to enable finding source files without support-levels defined.
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* Make non-normal dialplan execution routines be able to run on a hung up
channel. This is preparation work for hangup handler routines.
* Fixed ability to support relative non-normal dialplan execution
routines. (i.e., The context and exten are optional for the specified
dialplan location.) Predial routines are the only non-normal routines that
it makes sense to optionally omit the context and exten. Setting a hangup
handler also needs this ability.
* Fix Return application being able to restore a dialplan location
exactly. Channels without a PBX may not have context or exten set.
* Fixes non-normal execution routines like connected line interception and
predial leaving the dialplan execution stack unbalanced. Errors like
missing Return statements, popping too many stack frames using StackPop,
or an application returning non-zero could leave the dialplan stack
unbalanced.
* Fixed the AGI gosub application so it cleans up the dialplan execution
stack and handles the autoloop priority increments correctly.
* Eliminated the need for the gosub_virtual_context return location.
Review: https://reviewboard.asterisk.org/r/1984/
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This change fixes case-sensitivity for device-specific subscriptions such that
the technology identifier is case-insensitive while the remainder of the device
string is still case-sensitive. This should also preserve the original case of
the device string as passed in to the event system. CCSS is the only feature
affected as it is the only consumer of device-specific event subscriptions.
The second part of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after the fix to the
events system.
This adds a unit test to verify that the event system works as expected.
(closes issue ASTERISK-19422)
Review: https://reviewboard.asterisk.org/r/1780/
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This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated. This also adds
deprecation warnings for those features when used and in documentation.
Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)
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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/
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r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) | 5 lines
Be more explicit for CCSS generic device state event subscription.
Make CCSS generic device state event subscription specify the
AST_EVENT_IE_STATE ie exists to be safe.
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r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) | 21 lines
CCSS generic agent with POTS and ISDN phones fail caller busy call-back test.
If the following is true after a CCSS activation:
* The generic agent is for an analog phone or ISDN phone. (Caller party)
* The called party becomes available.
* The caller party is not available.
When the caller party becomes available, the caller is not alerted to the
called party being available. The generic agent still thinks the caller
is busy.
* Fixed the generic agent device state event subscription to look for all
device states that are considered available.
* Encapsulated the device state test for CCSS generic device available in
cc_generic_is_device_available(). Made the generic agent and monitor use
the new function instead of the manually coded inline equivalent.
JIRA AST-559
JIRA SWP-3462
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r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) | 13 lines
Deadlock between generic CCSS agent and native ISDN CCSS.
Deadlock can occur when the generic CCSS agent is deleting duplicate CC
offers and the native ISDN CC driver is processing an incoming CC message.
The cc_core_instances container lock cannot be held when an agent or
monitor callback is invoked without the possibility of a deadlock.
* Make kill_duplicate_offers() remove the reference in cc_core_instances
outside of the container lock.
JIRA AST-566
JIRA SWP-3469
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Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.
There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation. The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities. A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.
The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.
For example, you may have a single button that when not lit, there is no
active CCSS request. When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel(). If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful. The actual request could ultimately fail. Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.
The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary. The idea is to allow some level of
customization as to the phone's behavior.
As an example, you may want the BLF key to go solid once you have
requested a callback. You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback. You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.
Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine. You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.
You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states. For example, you
may have an extension 3000 that is currently associated with device
SIP/3000. You could then create a feature code for that extension that
may look something like:
exten => *823000,hint,ccss:sip/3000
You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.
(closes issue #18788)
Reported by: p_lindheimer
Patches:
ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski
Review: https://reviewboard.asterisk.org/r/1105/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
The CallCompletionRequest()/CallCompletionCancel() dialplan applications
exit nonzero on normal failure conditions. The nonzero exit causes the
dialplan to hangup immediately. The dialplan author has no opportunity to
report success/failure to the user.
* Made always return zero so the dialplan can continue.
* Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. Also
documented the values set.
* Reduced the warning about no core instance in CallCompletionCancel() to
a debug message. It is a normal event and should not be output at the
WARNING level.
(closes issue #18763)
Reported by: p_lindheimer
Patches:
ccss.patch uploaded by p lindheimer (license 558) Modified
Tested by: p_lindheimer, rmudgett
JIRA SWP-3042
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r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 Feb 2011) | 37 lines
No response sent for SIP CC subscribe/resubscribe request.
Asterisk does not send a response if we try to subscribe for call
completion after we have received a 180 Ringing. You can only subscribe
for call completion when the call has been cleared.
When we receive the 180 Ringing, for this call, its call-completion state
is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it
trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
Because this is an invalid state change, it just ignores the message. The
only state Asterisk will accept our subscribe message is in the
'CC_CALLER_OFFERED' state.
Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
the call by sending a CANCEL.
Asterisk should always send a response. Even if its a negative one.
The fix is to allow for the CCSS core to notify a CC agent that a failure
has occurred when CC is requested. The "ack" callback is replaced with a
"respond" callback. The "respond" callback has a parameter indicating
either a successful response or a specific type of failure that may need
to be communicated to the requester.
(closes issue #18336)
Reported by: GeorgeKonopacki
Tested by: mmichelson, rmudgett
JIRA SWP-2633
(closes issue #18337)
Reported by: GeorgeKonopacki
Tested by: mmichelson
JIRA SWP-2634
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r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb 2011) | 6 lines
Add a couple of useful channel variables for the CC recall macro.
CC_EXTEN and CC_CONTEXT will allow you to determine the channel
and context that will be called when the recall occurs.
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r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb 2011) | 9 lines
Rearrange a bit of code in the generic CC recall operation.
By waiting to call the callback macro after the CC_INTERFACES,
extension, priority, and context have been set, this information
can be accessed more easily within the callback macro.
Reported by Philippe Lindheimer.
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
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r303153 | rmudgett | 2011-01-20 14:31:20 -0600 (Thu, 20 Jan 2011) | 22 lines
Merged revision 303098 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
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r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines
CC_INTERFACES does not get built correctly with local channels.
If local channels are used with CCSS, CC_INTERFACES gets garbage prepended
to it so the CC recall fails. Also CC_INTERFACES gets "&(null)" appended
to it.
* Initialize the buffer to eliminate the prepended garbage.
* Filter out the empty interface strings to eliminate the latter.
* Added a diagnostic message if the CC_INTERFACES is ever empty.
JIRA ABE-2740
JIRA SWP-2848
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Previously, I had added the ast_sched_thread stuff that was a generic scheduler
thread implementation. However, if you used it, it required using different
functions for modifying scheduler contents. This patch reworks how this is
done and just allows you to optionally start a thread on the original scheduler
context structure that has always been there. This makes it trivial to switch
to the generic scheduler thread implementation without having to touch any of
the other code that adds or removes scheduler entries.
In passing, I made some naming tweaks to add ast_ prefixes where they were not
there before.
Review: https://reviewboard.asterisk.org/r/1007/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@299091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) | 7 lines
Separate call completion config parameter allocation and default initialization.
If you ever have a need to reset the call completion config parameters
to defaults, now you can.
And no Virginia, C++ idioms do not always work in C.
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The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Matt Jordan
George Joseph
Corey Farrell
Kinsey Moore
Scott Griepentrog
Mark Michelson
David M. Lee
Jason Parker
Richard Mudgett
Andrew Latham
Kevin P. Fleming
Terry Wilson
David Vossel
Russell Bryant
Leif Madsen