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- When you need to refer to 'variable XXX' outside a block, it needs
to be declared as '__block XXX', otherwise it will not be available with-
in the block, making updating that variable hard to do, and ast_free
lead to issues.
- Removed the #error message
because it creates complications when compiling external projects
against asterisk For example when using a different compiler than the
one used to compile asterisk. The warning/error should be generated
during the configure process not the compilation process
ASTERISK-24917
Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2
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This change modifies how the the output from a CLI command is sent
to a client over AMI.
Output from the CLI command is now sent as a series of zero-or-more
Output: headers.
Additionally, commands that fail to execute (eg: no such command,
invalid syntax etc.) now cause an Error response instead of Success.
If the command executed successfully, but the manager unable to
provide the output the reason will be included in the Message:
header. Otherwise it will contain 'Command output follows'.
Depends on a new version of starpy (> 1.0.2) that supports the new
output format.
See pull-request https://github.com/asterisk/starpy/pull/34
ASTERISK-24730
Change-Id: I6718d95490f0a6b3f171c1a5cdad9207f9a44888
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The "Add qualify_timeout processing and eventing" patch introduced
an issue where contacts that had qualify_frequency set to 0 were
showing Unavailable instead Unknown. This patch checks for
qualify_frequency=0 and create an "Unknown" contact_status
with an RTT = 0.
Previously, the lack of contact_status implied Unknown but since
we're now changing endpoint state based on contact_status, I've
had to add new UNKNOWN status so that changes could trigger the
appropriate contact_status observers.
ASTERISK-24977: #close
Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7
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When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be
called as a function. This causes a compile error with raw threadstorage as
it uses NULL for cleanup. This fix uses a macro that provides NULL when
DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);"
with "{};" when DEBUG_THREADLOCALS is enabled.
ASTERISK-24975 #close
Reported by: Ashley Sanders
Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402
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A potential problem that can arise is the following:
* Bob's phone is programmed to automatically forward to Carol.
* Carol's phone is programmed to automatically forward to Bob.
* Alice calls Bob.
If left unchecked, this results in an endless loops of call forwards
that would eventually result in some sort of fiery crash.
Asterisk's method of solving this issue was to track which interfaces
had been dialed. If a destination were dialed a second time, then
the attempt to call that destination would fail since a loop was
detected.
The problem with this method is that call forwarding has evolved. Some
SIP phones allow for a user to manually forward an incoming call to an
ad-hoc destination. This can mean that:
* There are legitimate use cases where a device may be dialed multiple
times, or
* There can be human error when forwarding calls.
This change removes the old method of detecting forwarding loops in
favor of keeping a count of the number of destinations a channel has
dialed on a particular branch of a call. If the number exceeds the
set number of max forwards, then the call fails. This approach has
the following advantages over the old:
* It is much simpler.
* It can detect loops involving local channels.
* It is user configurable.
The only disadvantage it has is that in the case where there is a
legitimate forwarding loop present, it takes longer to detect it.
However, the forwarding loop is still properly detected and the
call is cleaned up as it should be.
Address review feedback on gerrit.
* Correct "mfgium" to "Digium"
* Decrement max forwards by one in the case where allocation of the
max forwards datastore is required.
* Remove irrelevant code change from pjsip_global_headers.c
ASTERISK-24958 #close
Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23
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* changes:
res_pjsip: Add global option to limit the maximum time for initial qualifies
pjsip_options: Add qualify_timeout processing and eventing
res_pjsip: Refactor endpt_send_request to include transaction timeout
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Currently when Asterisk starts initial qualifies of contacts are spread out
randomly between 0 and qualify_timeout to prevent network and system overload.
If a contact's qualify_frequency is 5 minutes however, that contact may be
unavailable to accept calls for the entire 5 minutes after startup. So while
staggering the initial qualifies is a good idea, basing the time on
qualify_timeout could leave contacts unavailable for too long.
This patch adds a new global parameter "max_initial_qualify_time" that sets the
maximum time for the initial qualifies. This way you could make sure that all
your contacts are initialy, randomly qualified within say 30 seconds but still
have the contact's ongoing qualifies at a 5 minute interval.
If max_initial_qualify_time is > 0, the formula is initial_interval =
min(max_initial_interval, qualify_timeout * random(). If not set,
qualify_timeout is used.
The default is "0" (disabled).
ASTERISK-24863 #close
Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
Tested-by: George Joseph <george.joseph@fairview5.com>
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This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint. Only dynamic contact add/delete actions
update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.
This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...
1. A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message. The default is 3000ms. When the timer expires, the contact is
marked unavailable.
2. Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'. When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The
existing endpoint events are generated appropriately.
ASTERISK-24863 #close
Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>
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This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
Since we currently have no control over pjproject transaction timeout, this
patch pulls the pjsip_endpt_send_request function out of pjproject and into
res_pjsip/endpt_send_transaction in order to implement that capability.
Now when the transaction is initiated, we also schedule our own pj_timer with
our own desired timeout.
If the transaction completes before either timeout, pjproject cancels its timer,
and calls our tsx callback where we cancel our timer and run the app callback.
If the pjproject timer times out first, pjproject calls our tsx callback where
we cancel our timer and run the app callback.
If our timer times out first, we terminate the transaction which causes
pjproject to cancel its timer and call our tsx callback where we run the app
callback.
Regardless of the scenario, pjproject is calling the tsx callback inside the
group_lock and there are checks in the callback to make sure it doesn't run
twice.
As part of this patch ast_sip_send_out_of_dialog_request was created to replace
its similarly named private function. It takes a new timeout argument in
milliseconds (<= 0 to disable the timeout).
ASTERISK-24863 #close
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747
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This change adds the following:
1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked.
2. Unit tests for the query set implementation.
3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups.
For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV
are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A,
with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit
transport has been provided. Configured transports on the system are taken into account to
eliminate resolved addresses which have no hope of completing.
ASTERISK-24947 #close
Reported by: Joshua Colp
Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e
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This implements "weak" references. The weakproxy object is a real ao2 with
normal reference counting of its own. When a weakproxy is pointed to a normal
object they hold references to each other. The normal object is automatically
freed when a single reference remains (the weakproxy). The weakproxy also
supports subscriptions that will notify callbacks when it does not point
to any real object.
ASTERISK-24936 #close
Reported by: Corey Farrell
Change-Id: Ib9f73c02262488d314d9d9d62f58165b9ec43c67
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OPTIONAL_API has conditionals to define AST_OPTIONAL_API and
AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined.
Unfortunately this is inside the include protection block, so only the
first status of AST_API_MODULE is respected. For example res_monitor
is an optional API provider, but uses func_periodic_hook. This makes
func_periodic_hook non-optional to res_monitor.
This changes optional_api.h so that AST_OPTIONAL_API and
AST_OPTIONAL_API_ATTR is redefined every time the header is included.
ASTERISK-17608 #close
Reported by: Warren Selby
Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679
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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
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Add the .gitignore and .gitreview files to the asterisk repo.
NB: You can add local ignores to the .git/info/exclude file
without having to do a commit.
Common ignore patterns are in the top-level .gitignore file.
Subdirectory-specific ignore patterns are in their own .gitignore
files.
Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69
Tested-by: George Joseph
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With this patch, chan_pjsip/res_pjsip now sets the native formats to the
codecs negotiated by a call.
* The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native
formats to include all the negotiated audio codecs instead of only the
initial preferred audio codec and later the currently received audio
codec.
* The audio frame handling in channel.c:ast_read() is more streamlined and
will automatically adjust to changes in received frame formats. The new
policy is to remove translation and pass the new frame format to the
receiver except if the translation was to a signed linear format. A more
long winded version is commented in ast_read() along with some caveats.
* The audio frame handling in channel.c:ast_write() is more streamlined
and will automatically adjust any needed translation to changes in the
frame formats sent. Frame formats sent can change for many reasons such
as a recording is being played back or the bridged peer changed the format
it sends. Since it is a normal expectation that sent formats can change,
the codec mismatch warning message is demoted to a debug message.
* Removed the short circuit check in
channel.c:ast_channel_make_compatible_helper(). Two party bridges need to
make channels compatible with each other. However, transfers and moving
channels among bridges can result in otherwise compatible channels having
sub-optimal translation paths if the make compatible check is short
circuited. A result of forcing the reevaluation of channel compatibility
is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc
options take effect consistently now. It is unfortunate that these two
options are enabled by default and negate some of the benefits to the
changes in channel.c:ast_read() by forcing translation through signed
linear on a two party bridge.
* Improved the softmix bridge technology to better control the translation
of frames to the bridge. All of the incoming translation is now normally
handled by ast_read() instead of splitting any translation steps between
ast_read() and the slin factory. If any frame comes in with an unexpected
format then the translation path in ast_read() is updated for the next
frame and the slin factory handles the current frame translation.
This is the final patch in a series of patches aimed at improving
translation path choices. The other patches are on the following reviews:
https://reviewboard.asterisk.org/r/4600/
https://reviewboard.asterisk.org/r/4605/
ASTERISK-24841 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4609/
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This patch adds support for automatically detecting the type of DTMF that a
PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto',
the channel created for an endpoint will attempt to determine if RFC 4733
DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type
for the channel will be set to inband.
Review: https://reviewboard.asterisk.org/r/4438
ASTERISK-24706 #close
Reported by: yaron nahum
patches:
yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676)
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The NAPTR and SRV branches were worked on independently and
resulted in some code being duplicated in each. Since both
have been merged into trunk now, this patch reduces the
duplication by factoring out common code into its own
source files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This fixes autological comparison warnings in the following:
* chan_skinny: letohl may return a signed or unsigned value, depending on the
macro chosen
* func_curl: Provide a specific cast to CURLoption to prevent mismatch
* cel: Fix enum comparisons where the enum can never be negative
* enum: Fix comparison of return result of dn_expand, which returns a signed
int value
* event: Fix enum comparisons where the enum can never be negative
* indications: tone_data.freq1 and freq2 are unsigned, and hence can never be
negative
* presencestate: Use the actual enum value for INVALID state
* security_events: Fix enum comparisons where the enum can never be negative
* udptl: Don't bother to check if the return value from encode_length is less
than 0, as it returns an unsigned int
* translate: Since the parameters are unsigned int, don't bother checking
to see if they are negative. The cast to unsigned int would already blow
past the matrix bounds.
* res_pjsip_exten_state: Use a temporary value to cache the return of
ast_hint_presence_state
* res_stasis_playback: Fix enum comparisons where the enum can never be
negative
* res_stasis_recording: Add an enum value for the case where the recording
operation is in error; fix enum comparisons
* resource_bridges: Use enum value as opposed to -1
* resource_channels: Use enum value as opposed to -1
Review: https://reviewboard.asterisk.org/r/4533
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4533.patch submitted by dkdegroot (License 6600)
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operation.
When a channel enters the bridging system it is first made compatible with
the bridge and then the bridge technology makes the channel compatible
with the technology. For all but the DAHDI native and softmix bridge
technologies the make channel compatible with the bridge step is an
effective noop because the other technologies allow all audio formats.
For the DAHDI native bridge technology it doesn't matter because it is not
an initial bridge technology and chan_dahdi allows only one native format
per channel. For the softmix bridge technology, it is a noop at best and
harmful at worst because the wrong translation path could be setup if the
channel's native formats allow more than one audio format.
This is an intermediate patch for a series of patches aimed at improving
translation path choices.
* Removed code dealing with the unnecessary step of making the channel
compatible with the bridge.
ASTERISK-24841
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4600/
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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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This adds NAPTR record allocation and sorting, as well as
unit tests that verify that NAPTR records are parsed and
sorted correctly.
Review: https://reviewboard.asterisk.org/r/4542
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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A change in r430179 inserted a variable near the top of a
structure caused a problem when running DPMA in a version
of Asterisk compiled across the change. This patch moves
the new variable to the end of the structure, eliminating
the problem.
Review: https://reviewboard.asterisk.org/r/4574/
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This change adds support for parsing SRV records and consuming their values
in an easy fashion. It also adds automatic sorting of SRV records according
to RFC 2782.
Tests have also been included which cover parsing, sorting, and off-nominal
cases where the record is corrupted.
ASTERISK-24931 #close
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/4528/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Binary modules are sometimes built against the latest release of
Asterisk in each branch, and need to be compatible with all
releases of that branch. This change ensures that utils.h only
uses ast_log_safe from the core. For modules and utilities ast_log
is used instead.
Review: https://reviewboard.asterisk.org/r/4548/
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This patch fixes some invalid enum conversion warnings caught by clang. In
particular:
* chan_sip: Several functions mixed usage of the st_refresher_param
enum and st_refresher enum. This patch corrects the functions to use the
right enum.
* chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state.
* strings: Fixed incorrect usage of AO2 flags with strings container.
* res_stasis: Change a return enumeration to stasis_app_user_event_res.
Review: https://reviewboard.asterisk.org/r/4535
ASTERISK-24917
Reported by: dkdegroot
patches:
rb4535.patch submitted by dkdegroot (License 6600)
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Asterisk had an issue where retransmissions of MESSAGE requests resulted in
Asterisk processing the retransmission as if it were a new MESSAGE request.
This patch fixes the issue by creating a transaction in PJSIP on the incoming
request. This way, if a retransmission arrives, the PJSIP transaction layer
will resend the response and Asterisk will not ever see the retransmission.
ASTERISK-24920 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/4532/
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Use the standard ast_log instead of ast_log_safe for STANDALONE programs.
Review: https://reviewboard.asterisk.org/r/4538/
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This introduces a new logger routine ast_log_safe. This routine should be
used for all error messages in code that can be run as a result of ast_log.
ast_log_safe does nothing if run recursively. All error logging in
astobj2.c, strings.c and utils.h have been switched to ast_log_safe.
This required adding support for raw threadstorage. This provides direct
access to the void* pointer in threadstorage. In ast_log_safe, NULL is used
to signify that this thread is not already running ast_log_safe, (void*)1 when
it is already running. This was done since it's critical that ast_log_safe
do nothing that could log during recursion checking.
ASTERISK-24155 #close
Reported by: Timo Teräs
Review: https://reviewboard.asterisk.org/r/4502/
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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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Attempting to execute DTMF in a confbridge while file playback (prompt,
announcement, etc) is occurring is not allowed. You have to wait until
the sound file has completed before entering DTMF. This patch fixes it
so that app_confbridge now monitors for dtmf key presses during menu
driven file playback. If a key is pressed playback stops and it executes
the matched menu option.
ASTERISK-24864 #close
Reported by: Steve Pitts
Review: https://reviewboard.asterisk.org/r/4510/
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This change adds an abstracted core DNS API which resembles the API described
here[1]. The API provides a pluggable mechanism for resolvers and also a
consistent view for records. Both synchronous and asynchronous queries are
supported.
This change also adds a res_resolver_unbound module which uses the libunbound
library to provide resolution.
Unit tests have also been written for all of the above to confirm the API and
functionality.
ASTERISK-24834 #close
Reported by: Matt Jordan
ASTERISK-24836 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4474/
Review: https://reviewboard.asterisk.org/r/4512/
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API
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messages.
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
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Using DEBUG_CHAOS several instances of a null
pointer crash, and one uninitialized variable
were uncovered and fixed. Also added details
on why Asterisk failed to initialize.
Review: https://reviewboard.asterisk.org/r/4468/
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Locate potential crashes by exercising seldom
used code paths. This patch introduces a new
define DEBUG_CHAOS, and mechanism to randomly
return an error condition from functions that
will seldom do so. Functions that handle the
allocation of memory get the first treatment.
Review: https://reviewboard.asterisk.org/r/4463/
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Updated some documentation stating that endpoint identifiers registered without
a name are place at the front of the lookup list. Also renamed register method
'ast_sip_register_endpoint_identifier_by_name' to
'ast_sip_register_endpoint_identifier_with_name'
ASTERISK-24840
Reported by: Mark Michelson
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This patch fixes previously reverted code that caused binary incompatibility
problems with some modules. And like the original patch it makes sure that
no matter what order the endpoint identifier modules were loaded, priority is
given based on the ones specified in the new global 'endpoint_identifier_order'
option.
ASTERISK-24840
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4489/
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Due to a break in binary compatibility with some other modules these changes
are being reverted until the issue can be resolved.
ASTERISK-24840
Reported by: Mark Michelson
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Switch logger callid's from AO2 objects to simple integers.
This helps in two ways. Copying integers is faster than
referencing AO2 objects, so this will result in a small
reduction in logger overhead. This also erases the possibility
of an infinate loop caused by an invalid callid in
threadstorage.
ASTERISK-24833 #comment Committed callid conversion to trunk.
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4466/
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RAII_VAR, which is used extensively in Asterisk to manage reference counted
resources, uses a GCC extension to automatically invoke a cleanup function
when a variable loses scope. While this functionality is incredibly useful
and has prevented a large number of memory leaks, it also prevents Asterisk
from being compiled with clang.
This patch updates the RAII_VAR macro such that it can be compiled with clang.
It makes use of the BlocksRuntime, which allows for a closure to be created
that performs the actual cleanup.
Note that this does not attempt to address the numerous warnings that the clang
compiler catches in Asterisk.
Much thanks for this patch goes to:
* The folks on StackOverflow who asked this question and Leushenko for
providing the answer that formed the basis of this code:
http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang
* Diederik de Groot, who has been extremely patient in working on getting this
patch into Asterisk.
Review: https://reviewboard.asterisk.org/r/4370/
ASTERISK-24133
ASTERISK-23666
ASTERISK-20399
ASTERISK-20850 #close
Reported by: Diederik de Groot
patches:
RAII_CLANG.patch uploaded by Diederik de Groot (License 6600)
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Done as a separate commit from a finding in
https://reviewboard.asterisk.org/r/4467/
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When a type=global section is not defined in pjsip.conf the global
defaults are not applied. As a result the mandatory Max-Forwards header
is not added to SIP messages for res_pjsip/chan_pjsip.
The handling of pjsip.conf type=global objects has several problems:
1) If the global object is missing the defaults are not applied.
2) If the global object is missing the default_outbound_endpoint's default
value is not returned by ast_sip_global_default_outbound_endpoint().
3) Defines are needed so default values only need to be changed in one
place.
* Added a sorcery instance observer callback to check if there were any
type=global sections loaded. If there were more than one then issue an
error message. If there were none then apply the global defaults.
* Fixed ast_sip_global_default_outbound_endpoint() to return the
documented default when no type=global object is defined.
* Made defines for the global default values.
* Increased the default_useragent[] size because SVN version strings can
get lengthy and 128 characters may not be enough.
* Fixed an off-nominal code path ref leak in global_alloc() if the string
fields fail to initialize.
* Eliminated RAII_VAR in get_global_cfg() and
ast_sip_global_default_outbound_endpoint().
ASTERISK-24807 #close
Reported by: Anatoli
Review: https://reviewboard.asterisk.org/r/4467/
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A race condition happened between initiating a transfer and requesting
that a dialog termination be delayed. Occasionally, the transferrer
channels would exit the bridge and hangup before the dialog termination
delay was requested.
* Made request dialog termination delay before initiating the transfer
action. If the transfer fails then cancel the delayed dialog termination
request.
ASTERISK-24755 #close
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/4460/
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It's possible to have a scenario that will create a conflict between endpoint
identifiers. For instance an incoming call could be identified by two different
endpoint identifiers and the one chosen depended upon which identifier module
loaded first. This of course causes problems when, for example, the incoming
call is expected to be identified by username, but instead is identified by ip.
This patch adds a new 'global' option to res_pjsip called
'endpoint_identifier_order'. It is a comma separated list of endpoint
identifier names that specifies the order by which identifiers are processed
and checked.
ASTERISK-24840 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4455/
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Sending the following ARI commands caused Asterisk to crash if the JSON
body 'variables' object passes values of types other than strings.
POST /ari/channels
POST /ari/channels/{channelid}
PUT /ari/endpoints/sendMessage
PUT /ari/endpoints/{tech}/{resource}/sendMessage
* Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(),
ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and
ast_ari_endpoints_send_message_to_endpoint().
ASTERISK-24751 #close
Reported by: jeffrey putnam
Review: https://reviewboard.asterisk.org/r/4447/
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This patch adds a self-destruction option to the
dial api. The usefulness of this is mostly when
using async mode to spawn a separate thread used
to handle the new call, while the calling thread
is allowed to go on about other business.
The only alternative to this option would be the
calling thread spawning a new thread, or hanging
around itself waiting to destroy the dial struct
after completion.
Example of use (minus error checking):
struct ast_dial *dial = ast_dial_create();
ast_dial_append(dial, "PJSIP", "200", NULL);
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, "Echo");
ast_dial_option_global_enable(dial, AST_DIAL_OPTION_SELF_DESTROY, NULL);
ast_dial_run(dial, NULL, 1);
The dial_run call will return almost immediately
after spawning the new thread to run and monitor
the dial. If the call is answered, it is placed
into the echo app. When completed, it will call
ast_dial_destroy() on the dial structure.
Note that any allocations made to pass values to
ast_dial_set_user_data() or dial options must be
free'd in a state callback function on any of:
AST_DIAL_RESULT_UNASWERED,
AST_DIAL_RESULT_ANSWERED,
AST_DIAL_RESULT_HANGUP, or
AST_DIAL_RESULT_TIMEOUT.
Review: https://reviewboard.asterisk.org/r/4443/
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