This fixes build warnings found by GCC 8. In some cases format
truncation is intentional so the warning is just suppressed.
ASTERISK-27824 #close
Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84
This issue affected only installations with rtp_use_dynamic=yes in asterisk.conf
which is the default since Asterisk 15. Codec 2 and SiLK were built-in examples
of media formats which were affected.
ASTERISK-27850
Reported by: Dinis Brazão, Selene Feigl
Change-Id: I08c1e76433a67e4350141d38cacf3a1cb5086496
Due to a fixed size buffer the digest authentication could be
incorrectly calculated if a large URI was provided, causing
authentication failure. The buffer is now dynamically allocated to allow
any size URI within the normal limits of the HTTP request size.
ASTERISK-27841
Change-Id: I660609db13b8f9e5f9567f339dd804f4985d41b3
The stream topology has no lock of its own resulting in
another lock protecting it in some way (for example the
channel lock). If multiple channels are being juggled at
the same time this can be problematic. This change makes
the topology a reference counted object instead which
guarantees it will remain valid even without the channel
lock being held.
Change-Id: I4f4d3dd856a033ed55fe218c3a4fab364afedb03
How it works today:
media_cache tries to parse out the extension of the media file to be played
from the URI provided to Asterisk while caching the file.
What's expected:
Better will be to have Asterisk get extension from other ways too. One of the
common ways is to get the type of content from the CONTENT-TYPE header in the
HTTP response for fetching the media file using the URI provided.
Steps to Reproduce:
Provide a URL of the form: http://host/media/1234 to Asterisk for media
playback. It fails to play and logs show the following error line:
[Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c:
File http://host/media/1234 does not exist in any format
Scenario this issue is blocking:
In the case where the media files are stored in some cloud object store,
following can block the media being played via Asterisk:
Cloud storage generally needs authenticated access to the storage. The way
to do that is by using signed URIs. With the signed URIs there's no way to
preserve the name of the file.
In most cases Cloud storage returns a key to access the object and preserving
file name is also not a thing there
ASTERISK-27286
Reporter: Gaurav Khurana
Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89
If during translation a codec could not handle a given frame the translation
core would return NULL, thus not passing along the "missing" frame. Due to this
there was no frame to apply generic plc to, thus rendering it useless.
This patch makes it so the translation core produces an interpolated slin frame
in the cases where an attempt was made to translate to slin, but failed. This
interpolated frame is then passed along and can be used by the generic plc
algorithms to fill in the frame.
ASTERISK-27814 #close
Change-Id: I133d084da87adef913bf2ecc9c9240e3eaf4f40a
Redirect libc allocation functions to use Asterisk functions for
main/ast_expr2f.c and res/ael/ael_lex.c. This will resolve errors
produced by astmm.h when these files are regenerated, though other
issues still remain.
ASTERISK~27813
Change-Id: I7263e9e4217a17bde4ffaa2087a8f8aeb2a8588c
This change adds the ability for multiple REMB reports in
bridge_softmix to be combined according to a configured
behavior into a single report. This single report is sent
back to the sender of video, which adjusts the encoding bitrate
to be at or below the bitrate of the report. The available
behaviors are: lowest, highest, and average. Lowest uses the
lowest received bitrate. Highest uses the highest received
bitrate. Average goes through the received bitrates adding
them to the previous average and creates a new average.
Other behaviors can be added in the future and the existing
average one may be adjusted, but this provides the foundation
to do so.
Support for configuring which behavior to use has been
added to app_confbridge.
ASTERISK-27804
Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66
Replaces the never used opaque data array.
Updated stream tests to include get/set metadata and
stream clone with metadata.
Added stream metadata dump to "core show channel"
Change-Id: Id7473aa4b374d7ab53046c20e321037ba9a56863
Core bridging and, more specifically, bridge_softmix have been
enhanced to relay received frames of type TEXT or TEXT_DATA to all
participants in a softmix bridge. res_pjsip_messaging and
chan_pjsip have been enhanced to take advantage of this so when
res_pjsip_messaging receives an in-dialog MESSAGE message from a
user in a conference call, it's relayed to all other participants
in the call.
res_pjsip_messaging already queues TEXT frames to the channel when
it receives an in-dialog MESSAGE from an endpoint and chan_pjsip
will send an MESSAGE when it gets a TEXT frame. On a normal
point-to-point call, the frames are forwarded between the two
correctly. bridge_softmix was not though so messages weren't
getting forwarded to conference bridge participants. Even if they
were, the bridging code had no way to tell the participants who
sent the message so it would look like it came from the bridge
itself.
* The TEXT frame type doesn't allow storage of any meta data, such
as sender, on the frame so a new TEXT_DATA frame type was added that
uses the new ast_msg_data structure as its payload. A channel
driver can queue a frame of that type when it receives a message
from outside. A channel driver can use it for sending messages
by implementing the new send_text_data channel tech callback and
setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech
properties. If set, the bridging/channel core will use it instead
of the original send_text callback and it will get the ast_msg_data
structure. Channel drivers aren't required to implement this. Even
if a TEXT_DATA enabled driver uses it for incoming messages, an
outgoing channel driver that doesn't will still have it's send_text
callback called with only the message text just as before.
* res_pjsip_messaging now creates a TEXT_DATA frame for incoming
in-dialog messages and sets the "from" to the display name in the
"From" header, or if that's empty, the caller id name from the
channel. This allows the chat client user to set a friendly name
for the chat.
* bridge_softmix now forwards TEXT and TEXT_DATA frames to all
participants (except the sender).
* A new function "ast_sendtext_data" was added to channel which
takes an ast_msg_data structure and calls a channel's
send_text_data callback, or if that's not defined, the original
send_text callback.
* bridge_channel now calls ast_sendtext_data for TEXT_DATA frame
types and ast_sendtext for TEXT frame types.
* chan_pjsip now uses the "from" name in the ast_msg_data structure
(if it exists) to set the "From" header display name on outgoing text
messages.
Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489
Use of extended stringfields is a temporary mechanism to avoid ABI
breakage in released branches without resorting to more inconvienient
methods.
* Collect existing extended stringfields into the parent stringfield
section of the struct.
Change-Id: I8d46d037801b4518837c3ea4b6df95ceadc9436b
There is a problem when an INVITE-with-Replaces transfer targets a channel
in a ConfBridge. The transfer will unconditionally swap out the
ConfBridge channel. Unfortunately, the ConfBridge state will not be aware
of this change. Unexpected behavior will happen as a result since
ConfBridge channels currently can only be replaced by a masquerade and not
normal bridge channel moves.
* We just need to pretend that the channel isn't in a bridge (like other
transfer methods already do) so the transfer channel will masquerade into
the ConfBridge channel.
Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82
This change adds a configuration option to app_confbridge which can be
used to set the interval at which we will send a combined REMB (remote
estimated maximum bitrate) frame to sources of video. The bridging API
has also been extended slightly to allow setting this so bridge_softmix
can use it.
ASTERISK-27786
Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82
Adds a data buffer with a configurable size that can store different
kinds of packets (like RTP packets for retransmission). Given a number
it will store a data packet at that position relative to the others.
Given a number it will retrieve the given data packet if it is present.
This is purposely a storage of arbitrary things so it can be used not
just for RTP packets but also Asterisk frames in the future if needed.
The API does not internally use a lock, so it will be up to the user of
the API to properly protect the data buffer.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
Change-Id: Iff13c5d4795d52356959fe2a57360cd57dfade07
This change extends the existing AST_FRAME_RTCP frame type to be
able to contain additional RTCP message types, such as feedback
messages. The payload type is contained in the subclass which allows
knowing what is in the frame itself.
The RTCP feedback message type is now handled and REMB[1] messages
are raised with their containing information.
This also fixes a bug where all feedback messages were triggering
video updates instead of just FIR and FUR.
Finally RTCP frames are now passed up through the Asterisk core to
what is handling the channel, mapped appropriately in the case of
bridging, and written to an outgoing stream. Since RTCP frames are
on a per-stream basis this is only done on multistream capable
channels.
[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
ASTERISK-27758
ASTERISK-26366
Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e
Setting optind = 0 is forced to 1 in glibc implementation, but
causes option parsing to be flawed in other implementations, for
example on FreeBSD.
ASTERISK-27773 #close
Change-Id: Ia548e69f8302e9754dbbedb6bc451c0700c66f61
This creates a separate source to 'own' symbols related to options.h and
paths.h. This significantly reduces the number of exports created by
main/asterisk.o. This change is required to eventually be able to
link unmodified Asterisk sources to utilities and/or stand-alone tests.
ASTERISK~26245
Change-Id: I5cf184f4757f9363b80c9e678bdc35c477122380
* dahdi_chan_name
* dahdi_chan_name_len
* dahdi_chan_mode
* __manager_event
* dialed_interface_info
Added comment about __progname and environ being needed for FreeBSD to
prevent accidental removal in the future.
Change-Id: I3ae026bc541cd9cb572be2ffa95fc359547642b5
If the two formats on a channel are equal, we don't transcode and since
the generic plc needs slin to work, it doesn't get invoked.
* A new configuration option "genericplc_on_equal_codecs" was added
to the "plc" section of codecs.conf to allow generic packet loss
concealment even if no transcoding was originally needed.
Transcoding via SLIN is forced in this case.
ASTERISK-27743
Change-Id: I0577026a179dea34232e63123254b4e0508378f4
* Define CHAR_T_LIBEDIT and CHAR_TO_LIBEDIT based on
HAVE_LIBEDIT_IS_UNICODE. This avoids needing to repeatedly use
conditional blocks, eliminates having multiple function prototypes.
* Remove parenthesis from return values.
* Add missing code block brackets {}.
* Reduce use of 'else' conditional statements where possible.
Change-Id: I4315328ebea2f62641faf6881de2ac20a9f9d08e
This replaces AST_INLINE_API allocators in utils.h with real functions
implemented in astmm.c. Associated macro's are also moved from utils.h
to astmm.h.
Remove menuselect conflicts between MALLOC_DEBUG and DEBUG_CHAOS as they
can now be combined.
This has multiple benefits:
* Simplifies asterisk/utils.h by removing inline functions and use of
the logger.
* Removal of these inline functions decreases size of Asterisk and
module binaries by 1% or more.
* Puts memory management functions together with and without
MALLOC_DEBUG enabled, simplifying management of the code.
* Enables DEBUG_CHAOS for ASTMM_REDIRECT and bundled pjproject.
Change-Id: If9df4377f74bdbb627461b27a473123e05525887
* ast_cli_complete
* ast_complete_channels
* ast_complete_applications
These generators will now use ast_cli_completion_add if state == -1.
Change-Id: I7ff311f0873099be0e43a3dc5415c0cd06d15756
* acl (named_acl.c)
* cdr
* cel
* ccss
* dnsmgr
* dsp
* enum
* extconfig (config.c)
* features
* http
* indications
* logger
* manager
* plc
* sounds
* udptl
These modules are now loaded at appropriate time by the module loader.
Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so
the module loader will abort startup on failure of these modules.
Some of these modules are still initialized or shutdown from outside the
module loader. logger.c is initialized very early and shutdown very
late, manager.c is initialized by the module loader but is shutdown by
the Asterisk core (too much uses it without holding references).
Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f
GCC documentation states that when __attribute__((malloc)) is used it
should not return storage which contains any valid pointers. It
specifically mentions that realloc functions should not have the malloc
attribute, but this also means that complex initializers which could
contain initialized pointers should not use this attribute.
Change-Id: If507f33ffb3ca3b83b702196eb0e8215d27fc7d2
This causes MALLOC_DEBUG reporting to be slightly different, calls which
cause additional memory pools to be allocated now report the callers
location rather than the location which originally allocated the
string field structure. This reduces storage needed by string fields
and allows MALLOC_DEBUG to identify the source of additional allocations
rather than obscuring it by reporting the original allocation caller.
Change-Id: Idd18e6639a87ab862079b580c114d90361412289
When built-in components of Asterisk fail to start they cause the
Asterisk startup to abort. In these cases only the most critical
cleanup should be performed - closing databases and terminating
proceses. These cleanups are registered using ast_register_atexit, all
other cleanups should not be run during startup abort.
The main reason for this change is that these cleanup procedures are
untestable from the partially initialized states, if they fail it could
prevent us from ever running the critical cleanup with ast_run_atexits.
Create separate initialization for dns_core.c to be run unconditionally
during startup instead of being initialized by the first dns resolver to
be registered. This ensures that 'sched' is initialized before it can be
potentially used.
Replace ast_register_atexit with ast_register_cleanup in media_cache.c.
There is no reason for this cleanup to happen unconditionally.
Change-Id: Iecc2df98008b21509925ff16740bd5fa29527db3
In the script ./configure,
xyz_LIB is set by AST_PKG_CONFIG_CHECK and
xyz_LIBS is set by PKG_CHECK_MODULES within
AST_PKG_CONFIG_CHECK. Both are the same. In Asterisk normally the former and
only three times the latter was used. Let us use xyz_LIB without s, for
consistency with AST_EXT_LIB_CHECK. That eases understanding because now readers
do not have to know that xyz_LIB equals xyz_LIBS.
Change-Id: I7359860a5d730cdc784c2c48e501a082196434d3
Checking option_debug directly is incorrect as it ignores file/module
specific debug settings. This system-wide change replaces nearly all
direct checks for option_debug with the DEBUG_ATLEAST macro.
Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0
When HAVE_GETHOSTBYNAME_R_5 was set by the script ./configure, GCC 7.3.0 found
an unused variable. Actually, the variable was used (set to a dummy value) but
the compiler optimization might have removed that. Instead, this change ensures
that the variable 'res' is only used when it is really required.
Change-Id: Ic3ea23ccf84ac4bc2d501b514985b989030abab5
This allows asterisk to be compiled with MALLOC_DEBUG to load modules
built without MALLOC_DEBUG. Now pre-compiled third-party modules will
still work regardless of MALLOC_DEBUG being enabled or not.
Change-Id: Ic07ad80b2c2df894db984cf27b16a69383ce0e10
The pool cache gets in the way of finding use after free errors of memory
pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a
pool is released because it gets put into the cache instead of being
freed.
* Added the "cache_pools" option to pjproject.conf. Disabling the option
helps track down pool content mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the pool
contents are used after free and who freed it.
To disable the pool caching simply disable the cache_pools option in
pjproject.conf and restart Asterisk.
Sample pjproject.conf setting:
[startup]
cache_pools=no
* Made current users of the caching pool factory initialization and
destruction calls call common routines to create and destroy cached pools.
ASTERISK-27704
Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828
When a line is the maximum length "\n" is found at sizeof(buf) - 2 since
the last character is actually the null terminator. In addition if a
line was exactly 8190 plus a multiple of 8192 characters long the config
parser would skip the following line.
Additionally fix comment in voicemail.conf sample config. It previously
stated that emailbody can only contain up to 512 characters which is
always wrong. The buffer is normally 8192 characters unless LOW_MEMORY
is enabled then it is 512 characters. The updated comment states that
the line can be up to 8190 or 510 characters since the line feed and
NULL terminator each use a character.
ASTERISK-26688 #close
Change-Id: I80864a0d40d2e2d8cd79d72af52a8f0a3a99c015
When an RTP packet is received by an RTP engine it has to map the
payload into the Asterisk format. The code was incorrectly checking
our own static list for ALL payloads if it couldn't find a negotiated one.
This included dynamic payloads. If the payload mapped to a format
of a different type (for example receiving a video packet on an audio
RTP instance) then the core stream code could cause a crash if a legacy
channel driver was in use as no stream would be present.
To provide further protection the core stream code will no longer assume
that a video or audio frame will always have a stream for legacy channel
drivers. If no stream is present the frame is dropped.
ASTERISK-27488
Change-Id: I022556f524ad8379ee73f14037040af17ea3316a
This will make the source filename match the 'module reload sounds'
command. This will allow conversion to a built-in module in Asterisk 16
without needing to redefine AST_MODULE.
Change-Id: Ifb8e489575b27eb33d8c0b6a531f266670557f6e
Expand locking to include full reload process for extconfig to ensure
nothing can read the config mappings between clearing and reloading.
Change-Id: I378316bad04f1b599ea82d0fef62b8978a644b92
* Replace ad-hoc array management with macro's from vector.h.
* Remove redundent logger messages.
* Use normal Asterisk allocators instead of directly using libc
allocators.
* Free memory when an API has no implementation or users.
Change-Id: Ic6ecb31798d4a78e7df39ece86a68b60eac05bf5
Add an AMI events Load and Unload for notify when the
module has been loaded and unloaded.
ASTERISK-27661
Change-Id: Ib916c41eddd63651952998f2f49c57c42ef87a64
Jansson is thread safe for all read-only functions and reference
counting starting v2.11. This allows simplification of our code and
removal of locking around reference counting and dumping.
Change-Id: Id985cb3ffa6681f9ac765642e20fcd187bd4aeee
This removes the embedded copy of editline from the Asterisk source
tree, making a system copy of libedit mandatory in Asterisk 16+.
ASTERISK-27634 #close
Change-Id: Iedb64ad92acb78419f3caefedaa2bb7cd2a1a33f
Need to remove all CDR's listed by a CDR object from the active_cdrs_all
container including the root/master record.
ASTERISK-27656
Change-Id: I48b4970663fea98baa262593d2204ef304aaf80e
* Changed to create ami_event string only when the given blob is not
json_null().
* Fixed bad expression.
ASTERISK-27621
Change-Id: Ice58c16361f9d9e8648261c9ed5d6c8245fb0d8f
ast_str_append_event_header() could potentially leak and corrupt memory if
the ast_str needed to expand to add the AMI event header.
* Fixed to return error if the ast_str_append() failed.
Change-Id: I92f36b855540743b208d76e274152ee2d758176d
* Made not allocate memory if the channel snapshot is an internal channel.
* Free memory earlier when no longer needed.
Change-Id: Ia06e0c065f1bd095781aa3f4a626d58fa4d28b38
The dsp_talking_threshold does not represent time in milliseconds. It
represents the average magnitude per sample in the audio packets. This is
what the DSP uses to determine if a packet is silence or talking/noise.
Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443
This addresses all performance issues with 'module load' completion. In
addition to using ast_cli_completion_add we stop using libedit's
filename_completion_function, instead using ast_file_read_dir. This
ensures all results are produced from a single call to opendir.
Change-Id: I8bf51ffaa7ef1606f3bd1b5bb13f1905d72c6134
The previous fix broke the case
HAVE_SYSINFO = no
HAVE_SYSCTL = yes
HAVE_SWAPCTL = no
which occurs on FreeBSD 11.1 for example.
ASTERISK-26563
Change-Id: If77c39bc75f0b83a6c8a24ecb2fa69be8846160a
Move initialization of units which do not require configuration to occur
before preload modules. This leaves only units which load config between
module preload and regular load stages.
Change-Id: I1d15384acad16a22c3498124421af474fa517478
This change causes the configure script to throw an error if neither
__sync nor __atomic builtin functions are available.
ASTERISK-27619
Change-Id: Ie01a281e0f5c41dfeeb5f250c1ccea8752f56ef9
The code which handled loading modules had too many situations which
would result in halting Asterisk startup. Treat most errors as declines
instead of failures. The exception is when the module load function
returns AST_MODULE_LOAD_FAILURE or an invalid code.
Clear the missingdeps vector when appropriate to ensure the next loop
starts clean.
ASTERISK-27620
Change-Id: I45547d9641fd45bd86d80250224417625631ad84
* Copy more than one character at a time when there is nothing to
substitute.
* Fix off by one error if a '}' or ']' is missing.
* Eliminated the requirement that the "used" parameter had to point to a
variable. The current callers were always declaring a variable to meet
the requirement and discarding the value put into that variable. Now it
can be NULL.
* In ast_str_substitute_variables_full() fixed using the bogus channel to
evaluate a function. We were not using the bogus channel we just created
to help evaluate a subexpression.
Change-Id: Ia83d99f4f16abe47f329eb39b6ff2013ae7c9854
Each time the dial plan is reloaded, a lot of logs like these are generated:
"Added extension 'XXXXX' priority 1 to YYYYYYYYYYY"
This patch changes the log level for those logs.
ASTERISK-27084
Change-Id: I5662902161c50890997ddc56835d4cafb456c529
* Add SRC_EMBEDDED variable to main/Makefile. Built-in module sources
must be listed in this variable to ensure they get the correct CFLAGS.
Change-Id: I920852bc17513a9c2627061a4ad40511e3a20499
Use a single loop in a loop to scan the resource list attempting to
dlopen each module. The inner loop is repeated until it doesn't do any
work, then it is run one more time to allow printing of error messages.
Change-Id: I60c15cd57ff9680b62e2a94c7519401fa4a38e45
Dependency loader is now in place so we no longer need a separate loader
phase for global symbols only. This simplifies the loader and allows us
to minimize calls to dlopen.
Change-Id: I33e3174d67f3b4552d3d536326dcaf0ebabb097d
* Add string vectors for requires, optional_apis and enhances.
* Add reffed_deps module vector for holding references to dependencies.
* Initialize string vectors after final dlopen of each module.
* Free string vectors and clear references from reffed_deps in
module_destroy.
* Create functions necessary to process module dependencies and enforce
load order.
Module dependencies result in automatic references being managed by the
module loader. This enforces unload order.
Change-Id: I9be08d1dd331aceadc1dcba00b804d71360b2fbb
* Remove comment about lazy load.
* Improve message about module already being loaded and running.
* Handle allocation error in add_to_load_order.
* Dead code elimination from modules_shutdown.
Change-Id: I22261599c46d0f416e568910ec9502f45143197f
Since v12 the number of taskprocessors in the system has increased a lot.
Small systems can easily have over a hundred and larger systems can have
thousands.
Most uses of the tps_singletons container deal with creating and
destroying the taskprocessors. However, the pjsip distributor looks up
taskprocessors/serializers by name frequently. It needs to find the
serializer for incoming SIP responses to distribute them to the
appropriate serializer.
Change-Id: Ice0603606614ba49f7c0c316c524735c064e7e43
ast_vector_string_split:
This function will add items to an ast_vector_string by splitting values
of a string buffer. Items are appended to the vector in the order they
are found.
ast_vector_const_string:
A vector of 'const char *'.
Change-Id: I1bf02a1efeb2baeea11c59c557d39dd1197494d7
ast_format_get_sample_rate(.) returns an unsigned type. The difference of a
substraction between two unsigned types does not get implicitly converted to a
signed type. Therefore, using abs(.) did not make sense.
ASTERISK-27549
Change-Id: Ib904d9ee0d46b6fdd1476fbc464fbbf813304017
pbx_extension_helper has a check for q->swo.exec == NULL but it doesn't
actually return so we would still run the function. Fix the return.
Move the 'int res' variable into the only scope which uses it.
Also fix a copy-paste error in ast_pbx_init which could result in a
crash on allocation failure (we exit with a normal error instead).
Change-Id: I0693af921fdc7f56b6a72a21fb816ed08b960a69
Translators are run during module load before the module is actually
running, so it cannot use ast_module_running_ref.
ASTERISK-20346
Change-Id: Iaa0e75da99c696e38000f1a41e340abbd7a88f56
This function returns NULL if the module in question is not running. I
did not change ast_module_ref as most callers do not check the result
and they always call ast_module_unref.
Make use of this function when running registered items from:
* app_stack API's
* bridge technologies
* CLI commands
* File formats
* Manager Actions
* RTP engines
* Sorcery Wizards
* Timing Interfaces
* Translators
* AGI Commands
* Fax Technologies
ASTERISK-20346 #close
Change-Id: Ia16fd28e188b2fc0b9d18b8a5d9cacc31df73fcc
Per RFC 5245, the foundation specified with an ICE candidate can be up
to 32 characters but we are only allowing for 31.
ASTERISK-27498 #close
Reported by: Michele Prà
Change-Id: I05ce7a5952721a76a2b4c90366168022558dc7cf
This uses AO2_STRING_FIELD_HASH_FN and AO2_STRING_FIELD_CMP_FN where
possible in the Asterisk core.
This removes CMP_STOP from the result of CMP_FN callbacks for the
following structure types:
* ast_bucket_metadata
* ast_bucket_scheme
* generic_monitor_instance_list (ccss.c)
* ast_bucket_file (media_cache.c)
* named_acl
Change-Id: Ide4c1449a894bce70dea1fef664dade9b57578f1
Add a reference to the calling module when it is active to protect
access to datastore->info. Remove module references done by
func_periodic_hook as the datastore now handles it.
ASTERISK-25128 #close
Change-Id: I8357a3711e77591d0d1dd8ab4211a7eedd782c89
* Use current OBJ_SEARCH_xxx defines instead of the deprecated versions.
* Fix hash_cb and cmp_cb container functions to correctly use the
OBJ_SEARCH_xxx values.
* Remove incorrect usage of CMP_STOP. Most uses in the system have no
effect. This allows the collapse of channel_role_single_cmp_cb() and
channel_role_multi_cmp_cb() into channel_role_cmp_cb().
* Remove unnecessary usage of RAII_VAR().
Change-Id: I02c405518cab22aa2a082b61e2353bf7cd629a70
The AMI Status event had linkedid listed twice and was missing the
effective connected line name and number headers.
NOTE: The linkedid and other standard channel snapshot fields in the XML
documentation are part of the <channel_snapshot/> XML template defined in
doc/appdocsxml.xslt.
Change-Id: I004c4c4f9e7b40ef55035c831702721bec82496c
* handle_dial_message: Missing a check for NULL peer.
* cdr_generic_register: Missing unlock on allocation failure.
cdr_generic_register is fixed by reordering so the new structure is
allocated and initialized before locking the list.
Change-Id: I5799b99270d1a7a716a555c31ac85f4b00ce8686
Some compiler optimizers seem to assume that dlopen will not use
__attribute__((constructor)) functions to call back to the program.
This was causing resource_being_loaded to be optimized away completely.
ASTERISK-27531 #close
Tested By: abelbeck
Change-Id: If17a3b889e06811a0e7119f0539d052494d6ece9
Fix instances of:
* Retreive
* Recieve
* other then
* different then
* Repeated words ("the the", "an an", "and and", etc).
* othterwise, teh
ASTERISK-24198 #close
Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31
The bridge holds onto the old channel video source after it's been
released. This can lead to use after free errors.
ASTERISK-27229 #close
Change-Id: Ib2dab61677dd8a21f7ad53cdc9b8ca93297838b3
Apparently in OSX it's possible for OSX to HAVE_SYSCTL but not
HAVE_SYSINFO or HAVE_SWAPCTL. In this case freeswap caused an unused
variable error.
ASTERISK-26563 #close
Change-Id: I8ec5b1897b786cc1abaf62264aa75039eea05510
Corey Farrell
Alexander Traud
Jaco Kroon
Jenkins2
Joshua Colp
Sean Bright
Gaurav Khurana
George Joseph
Richard Mudgett
Kevin Harwell
Ben Ford
Florian Floimair
Guido Falsi
Sungtae Kim
krells
Tzafrir Cohen