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
* listen uses the variable `s` for the result from ast_poll() then
overwrites it with the result of accept(). Create a separate variable
poll_result to avoid confusion since ast_poll does not return a file
descriptor.
* Resolve fd leak that would occur if setsockopt failed in listen.
* Reserve an extra byte while processing completion results from remote
daemon. This fixes a bug where completion processing used strstr() on
a string that was not '\0' terminated. This was no risk to the Asterisk
daemon, the bug was only reachable the remote console process.
* Resolve leak in handle_showchan when the channel is not found.
* Multiple leaks and a deadlock in pbx_config CLI completion.
* Fix leaks in "manager show command".
Change-Id: I8f633ceb1714867ae30ef4e421858f77c14485a9
Add a check to allocate_dns_record to prevent calling a pointer
retrieved from beyond dns_alloc_table.
ASTERISK-27495 #close
Change-Id: Ie2f6e4991cea46baa12e837bd64cc22b44d322bb
When a channel that is on hold gets added to a bridge by
the Bridge AMI action or the dialplan application of the same name,
music continues to play, causing "robotic sound".
This commit adds a call to ast_moh_stop to stop the music.
Also, it makes the AMI Park action use the right MOH class when the
channel gets parked.
Reported by: Zane Conkle
ASTERISK-25079 #close
Change-Id: I4b129c5a20c15e63968842460ac5a1a85903cf9f
Some variables are set and never changed, making them constant. This
means that code in the 'false' block of the conditional is unreachable.
In chan_skinny and res_config_ldap I used preprocessor directive `#if 0`
as I'm unsure if the unreachable code could be enabled in the future.
Change-Id: I62e2aac353d739fb3c983cf768933120f5fba059
* Fix small leaks in from error conditions in sdp.c and translate.c.
* Check new file descriptor is less than 0, not less than or equal.
Change-Id: Id7782775486175c739e0c4bf3ea5e17e3f452a99
* ast_linear_stream would leak a file descriptor if it failed to allocate
lin.
* ast_control_tone leaked zone and ts if ast_playtones_start failed.
Additionally added whitespace to ast_linear_stream, pulled assignments
out of conditionals for improved readability.
Change-Id: I6d1a10cf9161b1529d939b9b2d63ea36d395b657
This moves netsock.c / netsock.h to the chan_iax2 module. netsock.h has
been marked deprecated since 13.0.0, chan_iax2 is the only remaining
user.
Change-Id: I28c6578043bac18de5ea608e136acec4f83d5dd3
The completion generator is missing a return so typing "core set debug
all off <tab>" causes the command to actually execute.
Change-Id: Ibf6462088a74eee66967732b50445783ebefc20b
Remove nearly all use of regex from ACO users. Still remaining:
* app_confbridge has a legitamate use of option name regex.
* ast_sorcery_object_fields_register is implemented with regex, all
callers use simple prefix based regex. I haven't decided the best
way to fix this in both 13/15 and master.
Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b
ACO uses regex in many situations where it is completely unneeded. In
some cases this doubles the total processing performed by
aco_process_config.
* Create ACO_IGNORE category type for use in place of skip_category
regex source string.
* Create additional aco_category_op values to allow specifying category
filter using either a single plain string or a NULL terminated array
of plain strings.
* Create ACO_PREFIX to allow matching option names to case insensitive
prefixes.
Change-Id: I66a920dcd8e2b0301f73f968016440a985e72821
This is needed for future changes which will require being able to
process the load priority out of order.
Change-Id: Ia23421197f09789940510b03ebbbf3bf24d51bea
* Split off load_dlopen to perform actual dlopen, check results and log
warnings when needed.
* Always use RTLD_NOW.
* Use flags which minimize number of calls to dlopen required. First
attempt always uses RTLD_GLOBAL when global_symbols_only is enabled,
RTLD_LOCAL when it is not.
This patch significantly reduces the number of dlopen's performed. With
299 modules my system ran dlopen 857 times before this patch, 655 times
after this patch.
Change-Id: Ib2c9903cfddcc01aed3e01c1e7fe4a3fb9af0f8b
This protects the module loader itself against crashing if dlopen is
called on a module from outside loader.c.
* Expand scope of lock inside ast_module_register to include reading of
resource_being_loaded.
* NULL check resource_being_loaded.
* Set resource_being_loaded NULL as soon as dlopen returns. This fixes
some error paths where it was not NULL'ed.
* Create module_destroy function to deduplicate code from
ast_module_unregister and modules_shutdown.
* Resolve leak that occured if a module did not successfully register.
* Simplify checking for successful registration.
Change-Id: I40f07a315e55b92df4fc7faf525ed6d4f396e7d2
We should not do flood detection on video RTP streams. Video RTP streams
are very bursty by nature. They send out a burst of packets to update the
video frame then wait for the next video frame update. Really only audio
streams can be checked for flooding. The others are either bursty or
don't have a set rate.
* Added code to selectively disable packet flood detection for video RTP
streams.
ASTERISK-27440
Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70
rasterisk does not need to handle setting verbose levels locally, it
should just tell the daemon what it wants and print what it is given.
Just max out the verbose level on the local client so all filtering
happens on the daemon.
ASTERISK-20281 #close
Change-Id: Ia305f75f1fc424a9169bfa30ef70d626ace2c8a8
A couple of places were setting the status to "UNKNOWN" when qualifies were
being disabled. Instead this should be set to the "CREATED" status that
represents when a contact is given (uri available), but the qualify frequency
is set to zero so we don't know the status.
This patch updates the relevant places with "CREATED". It also updates the
"CREATED" status description (value shown in CLI/AMI/ARI output) to a value
of "NonQualified"/"NonQual" as this description is hopefully less confusing.
ASTERISK-27467
Change-Id: Id67509d25df92a72eb3683720ad2a95a27b50c89
Using the LIKE operator requires a full table scan of 'astdb', whereas a
comparison operation is able to use the primary key index.
This patch adds a new function to the AstDB API for quick prefix matches
and updates res_sorcery_astdb to utilize it. This showed substantial
performance improvement in my test environment.
Related to ASTERISK~26806, but does not completely resolve it.
Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1
Optimize resource_name_match. This change eliminates use of
ast_strdupa, instead verifying that both basename's are the same length,
then using strncasecmp.
Change-Id: I477275c0e954c99d74be5abfc8bb6545b04e5a3d
There are many places in the code base where we ignore the return value
of fcntl() when getting/setting file descriptior flags. This patch
introduces a convenience function that allows setting or clearing file
descriptor flags and will also log an error on failure for later
analysis.
Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7
The sounds index is rebuilt each time a format is registered or
unregistered. This causes the index to be repeatedly rebuilt during
startup and shutdown.
This patch significantly reduces the work done by delaying sound index
initialization until after modules are loaded. This way a reindex only
occurs if a format module is loaded after startup. We also skip
reindexing when format modules are unloaded during shutdown.
Change-Id: I585fd6ee04200612ab1490dc804f76805f89cf0a
This eliminates some wasteful operations in media_index startup.
* Replace statically set string-fields with char[0].
* Eliminate pointless RAII_VAR's.
* alloc_variant: Avoid pointless ao2_find on new info->variant.
* Stop trying find_variant before alloc_variant.
* process_media_file: replace ast_str with ast_asprintf. This avoids
reallocation of file_id_str.
Overall sounds_index.c is about 27% of Asterisk startup time when using
sample configs. This patch reduces it to 20%. This is a half-fix. The
real problem is that the media_index is regenerated repeatedly - 68
times in my test.
Change-Id: Ia50b752f8efb356f852b05c4be495a6631af8652
* Added start DTMF transfer verbose messages.
* Made associated transfer messages use a similar message format.
* Adjusted message verbose level as requested by initial reporter.
ASTERISK-27449
Change-Id: I2045714586414b3c5ef1f3cc56c1c4af4b31f551
* Add the channel name to diagnostic messages so you will know which
channel failed to transfer.
* Promoted some debug messages to verbose 4 messages.
ASTERISK-27449 #close
Change-Id: Idac66b7628c99379cc9269158377fd87dc97a880
ast_category_get() has an (undocumented) implementation detail where it
tries to match the category name first by an explicit pointer comparison
and if that fails falls back to a normal match.
When initially building an ast_config during ast_config_load, this
pointer comparison can never succeed, but we will end up iterating all
categories twice. As the number of categories using a template
increases, this dual looping becomes quite expensive. So we pass a flag
to category_get_sep() indicating if a pointer match is even possible
before trying to do so, saving us a full pass over the list of current
categories.
In my tests, loading a file with 3 template categories and 12000
additional categories that use those 3 templates (this file configures
4000 PJSIP endpoints with AOR & Auth) takes 1.2 seconds. After this
change, that drops to 22ms.
Change-Id: I59b95f288e11eb6bb34f31ce4cc772136b275e4a
When starting Asterisk in the foreground, there is a perceptible delay
when loading modules that use the ACO and sorcery config frameworks.
For example, a lightly configured res_pjsip took 853ms to load on my
VM.
I tracked down the slowness to the XPath queries used to associate the
relevant documentation with the config options. One improvement was
adding a call to xmlXPathOrderDocElems after loading an XML document.
From the libxml2 docs:
Call this routine to speed up XPath computation on static documents.
The second change was to remove recursive descent and wildcard
operators from the XPath queries. After these changes, res_pjsip takes
85ms to load on my VM and there is no longer a perceptible delay when
starting Asterisk in the foreground.
Change-Id: I45d457f1580e26bf5a2b0dab16e8e9ae46dcbd82
Previous commits maintained compatibility with older remote console
clients as well as maintaining all API's.
Remove the following compatibility code:
* ast_cli_generatornummatches.
* Remote command "_command nummatches".
* Sorting / duplicate removal by remote console.
Change-Id: I59e6ce94fa57ae564888442049695f7e46746437
When a format has no pre-recorded sound files, Asterisk has to transcode between
formats. For this, Asterisk has a fixed translation table. If the pre-recorded
sound files are not available in the same sample rate, Asterisk has not only to
transcode but also to resample.
Asterisk has pre-recorded files for SLN (8000 kHz) and SLN16 (16000 kHz).
However before this change, Asterisk did not take the sample rate into account,
because the translation paths to SLN and SLN16 got the same score/weight in the
table. Consequently, you might have got narrow-band audio with siren14, speex32,
silk24, and silk12 although those are (ultra) wide-band audio codecs.
With this change, the distance in sample-rates is taken into account. Now on the
Command-Line interface (CLI) 'core show channels', you should see:
(slin@16000)->(slin@32000)->(speex@32000).
ASTERISK-23735
Reported by: Richard Kenner
Change-Id: I9448295c1978be26f8633b6066395e7bbbe2e213
* Stop using "_COMMAND NUMMATCHES" on remote consoles. Using this
command had doubled the amount of work needed from the Asterisk
daemon for each completion request.
* Fix code formatting.
* Remove static buffer used to send the command, use the same buffer
that will receive the results.
* Move sort from ast_cli_display_match_list.
Change-Id: Ie2211b519a3d4bec45bf46e0095bdd01d384cb69
This rewrites ast_el_strtoarr to use vector's internally, but still
return the original NULL terminated array of strings.
Change-Id: Ibfe776cbe14f750effa9ca360930acaccc02e957
* Stop estimating line count, just print until we run out of matches.
* Stop freeing entries, the caller does that anyways.
* Stop calculating / returning numoutput, it was ignored.
Change-Id: I7f92afa8bea92241a95227587367424c8c32a5cb
Some completion generators are very inefficent due to the way CLI
requests matches one at a time. ast_cli_completion_add can be called
multiple times during one invokation of a CLI generator to add all
results without having to reinitialize the search state for each match.
Change-Id: I73d26d270bbbe1e3e6390799cfc1b639e39cceec
The ability to add to localized storage cannot be supported by
ast_cli_generator. The only calls to ast_cli_generator should be by
functions that need to proxy the CLI generator, for example 'cli check
permissions' or 'core show help'.
* ast_cli_generatornummatches now retrieves the vector of matches and
reports the number of elements (not including 'best' match).
* test_substitution retrieves and iterates the vector.
Change-Id: I8cd6b93905363cf7a33a2d2b0e2a8f8446d9f248
Make the comments follow doxygen format, move comments to the line
before each field they describe.
Change-Id: Ic445468398b5e88f13910f7c2f70bd15aad33a27
* Fix conditional in libasteriskssl.
* Use variables produced by configure to link the SSL and uuid libraries
into libasteriskpj.so instead of hard-coding them.
ASTERISK-27431
Change-Id: I3977931fd3ef8c4e4376349ccddb354eb839b58d
This adds the printf attribute and changes 'fmt' from 'const void *' to
'const char *'. This resolves a warning from some compiler for
vsnprintf needing a literal string for format.
Change-Id: I71c33a8262590042ee451e1146760c10bb22fb78
Add checks for allocation errors, cleanup and report failure when they
occur.
* ast_duplicate_acl_list: Replace log warnings with errors, add missing
line-feed.
* ast_append_acl: Add missing line-feed to logger message.
* ast_append_ha: Avoid ast_strdupa in loop by moving debug message to
separate function.
* ast_ha_join: Use two separate calls to ast_str_append to avoid using
ast_strdupa in a loop.
Change-Id: Ia19eaaeb0b139ff7ce7b971c7550e85c8b78ab76
This is a fun one.
Given the following attended transfer scenario:
1. Transfer target is called
2. Transferer hangs up
3. Transfer target call attempt reaches timeout
4. Transfer target is told to hang up
5. Transfer target answers before channel is hung up
6. Transferer recall target is called
A crash would occur. This is because the transfer target call
attempt, despite being told to hang up, would raise a recall
target answer before the recall target had been answered. As it
had not answered there would be no recall target channel and it
would implode.
This change makes it so that if the transfer target has been
hung up we don't tell the attended transfer code that it has
answered. We also clear out the stimulus that the recall target
has been answered after telling the transfer target to hang up,
in case it was able to raise the information before we told it
to hangup.
ASTERISK-27361
Change-Id: Ifb8b255a9c4d2c5c1b8ad77bf54f659ed286df99
Memory corruption happened to the media frame caches when an audio hook
freed a frame when it shouldn't. I think the freed frame was because a
jitter buffer interpolated a missing frame and the audio hook
unconditionally freed it.
* Made audiohook.c:audio_audiohook_write_list() not free an interpolated
frame if it is the same frame as what was passed into the routine.
* Made plc.c:normalise_history() use memmove() instead of memcpy() on a
memory block that could overlap. Found by valgrind investigating this
issue.
ASTERISK-27238
ASTERISK-27412
Change-Id: I548d86894281fc4529aefeb9f161f2131ecc6fde
Reduce the signal monitoring thread file descriptor use from two to one
on systems that support eventfd.
Change-Id: Id4041a237d481ff699639e153ea6982fee14a462
The remote console socket path is the combination of asterisk.conf
settings astrundir from [directories] and astctl from [files].
Unconditionally combine the two strings after processing all values
to ensure we end up with the correct socket path.
ASTERISK-27415
Change-Id: Ib1e2805d55d6b0955c6430a1a2a93acbf9b091e8
Some consumers of the sorcery API use ast_sorcery_retrieve_by_regex
only so that they can anchor the potential match as a prefix and not
because they truly need regular expressions.
Rather than using regular expressions for simple prefix lookups, add
a new operation - ast_sorcery_retrieve_by_prefix - that does them.
Change-Id: I56f4e20ba1154bd52281f995c27a429a854f6a79
This is a rewrite of ast_cli_completion_matches using a vector to build
the list. The original function calls the vector version, NULL
terminates the vector and extracts the elements array.
One change in behavior the results are now sorted and deduplicated. This
will solve bugs where some duplicate checking was done before the list
was sorted.
Change-Id: Iede20c5b4d965fa5ec71fda136ce9425eeb69519
The media frame cache gets in the way of finding use after free errors of
media frames. Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.
* Added the "cache_media_frames" option to asterisk.conf. Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the frame is
used after free and who freed it. NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.
To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.
Sample asterisk.conf setting:
[options]
cache_media_frames=no
ASTERISK-27413
Change-Id: I0ab2ce0f4547cccf2eb214901835c2d951b78c00
This adds FD tracking for the following functions:
* eventfd
* timerfd_create
* socketpair
* accept
ASTERISK-27404
Change-Id: Id6848fe904ade2d34eb39d2a20bd6b223e1111fc
cdr_object_update_party_b_userfield_cb() could overrun the fixed buffer if
the supplied string is too long. The long string could be supplied by
external means using the CDR(userfield) function.
This may seem reminiscent to AST-2017-001 (ASTERISK_26897) and it is. The
earlier patch fixed the buffer overrun for Party A's userfield while this
patch fixes the same thing for Party B's userfield.
ASTERISK-27337
Change-Id: I0fa767f65ecec7e676ca465306ff9e0edbf3b652
ast_stream_topology_set_stream had suppressed error codes from
AST_VECTOR_APPEND. The result of AST_VECTOR_APPEND needs to be returned
to the caller so they can take appropriate action on the stream.
Change-Id: I6c0d12755743eadba1357f6153526cc055592856
Message tech and handler registrations use a vector which could fail to
expand. If it does log and error and return error.
Change-Id: I593a8de81a07fb0452e9b0efd5d4018b77bca6f4
format_cap_framed_init can fail on AST_VECTOR_APPEND. This should
report failure to the caller and clean the newly allocated frame.
Change-Id: Ica0661235bf09497bf23d844ceb01f21b41a55b0
The internal CLI command "_command complete" was last used by Asterisk
0.2.0. Since then we've been using "_command nummatches" and "_command
matchesarray".
Change-Id: I682fe1e21a24a3bb5bd04146e639f1c5866bcfce
When (v)asprintf() fails, the state of the allocated buffer is undefined.
The library had better not leave an allocated buffer as a result or no one
will know to free it. The most likely way it can return failure is for an
allocation failure. If the printf conversion fails then you actually have
a threading problem which is much worse because another thread modified
the parameter values.
* Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL
on failure. That is much more useful than either an uninitialized pointer
or a pointer that has already been freed. Many uses won't have to check
for failure to ensure that the buffer won't be double freed or prevent an
attempt to free an uninitialized pointer.
* stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by
ast_asprintf().
* ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to
the wrong thing which is now not needed even if assigning to the right
thing.
Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23
This mimics the behavior of Chrome and Firefox and creates an ephemeral
X.509 certificate for each DTLS session.
Currently, the only supported key type is ECDSA because of its faster
generation time, but other key types can be added in the future as
necessary.
ASTERISK-27395
Change-Id: I5122e5f4b83c6320cc17407a187fcf491daf30b4
Asterisk can be compiled without a SSL/TLS library, without the Development
Headers of OpenSSL. However, if TLS (SIP) or Secure-WebSockets (WebRTC) was
enabled in a configuration file, Asterisk did not notice the user. Asterisk
failed silently, only the corresponding TCP ports were not open.
ASTERISK-27394
Reported-by: mossley74
Change-Id: Ib8b7539a5b2af8154c22e5f7a40fc68f95d95b93
We use the editline library to help with filename completion in our CLI
interface. Some systems failed to find the header when included from
loader.c. This is fixed by setting the proper CFLAGS for the build of
loader.o.
ASTERISK-27378
Change-Id: Ib7fd496f1d7ed48141a2eadd5dd61cab2f2308be
When a frame is provided to ast_write ensure that a multistream
capable channel has a stream for it before attempting to give it
to the channel driver. In some cases (such as a deferred SDP
negotiation) the stream may not yet exist.
ASTERISK-27364
Change-Id: Icf84ca982a67cdd6e9a71851eb7eb1bd0e865276
Replace 'needsreload' argument with a 'type' argument to specify which
type of modules you want completion. This provides more accurate CLI
completion for load and unload commands.
* 'module unload' now excludes modules that have active references or are
not running.
* 'module load' now excludes modules that are already running.
* 'core set debug [atleast] <level> [module]' shows running modules only.
ASTERISK-27378
Change-Id: Iea3e00054461484196c46f688f02635cc886bad1
The dialplan application "Bridge" was not setting the BRIDGERESULT to failure
when a failure did occur. Even worse if it did fail to join the bridge it would
still report success.
This patch now sets the BRIDGERESULT variable to an appropriate value for a
given condition state. Also, removed the value INCOMPATIBLE as a valid result
type since it is no longer used.
ASTERISK-27369 #close
Change-Id: I22588e7125a765edf35cff28c98ca143e9927554
In WebRTC streams (or media tracks in their world) can be grouped
together using the mslabel. This informs the browser that each
should be synchronized with each other.
This change extends the stream API so this information can
be stored with streams. The PJSIP support has been extended
to use the mslabel to determine grouped streams and store
this association on the streams. Finally when creating the
SDP the group information is used to cause each media stream
to use the same mslabel.
ASTERISK-27379
Change-Id: Id6299aa031efe46254edbdc7973c534d54d641ad
* Stop using ast_module_helper to check if a module is loaded, use
ast_module_check instead (app_confbridge and app_meetme).
* Stop ast_module_helper from listing reload classes when needsreload
was not requested.
ASTERISK-27378
Change-Id: Iaed8c1e4fcbeb242921dbac7929a0fe75ff4b239
Earlier versions of the codec_opus samples_count callback can return
negative error values on undecodable frames. This resulted in a divide by
zero exception.
* Added a defensive check in ast_codec_samples_count() for a "negative"
samples count return value. Log the event and set the count to zero.
ASTERISK-27194
Change-Id: Icf69350307ecbbc80a3d74de46af9bd80ea17819
Currently ast_http_send barricades a portion of the content that
needs to be sent in order to establish a connection for things
like the ARI client. The conditional and contents have been changed
to ensure that everything that needs to be sent, will be sent.
ASTERISK-27372
Change-Id: I8816d2d8f80f4fefc6dcae4b5fdfc97f1e46496d
The configure option to disable XML documentation does not currently
work. This patch makes it effective, but also causes an ABI change by
removing the ast_xmldoc_* symbols. Disabling xmldoc also prevents docs
from being automatically generated, but they can still be manually
generated with 'make doc/core-en_US.xml'.
ASTERISK-26639
Change-Id: Ifac562340c09f80c83e0203de098fcac93bf8c44
This makes the 'bt' parameter unconditional for ast_store_lock_info and
ast_remove_lock_info. The 'bt' parameter is unused when HAVE_BKTR is
undefined.
Change-Id: Ieced0e920928b735a39c3b5952b806c473d67453
A few places in hashtab use free instead of ast_free, remove declaration
of ASTMM_LIBC from hashtab.c as it's no longer needed.
Change-Id: I2ff089bad71640c03c3ce97f1b00fc962ef79427
* Rename the Party A CDR container from active_cdrs_by_channel to
active_cdrs_master.
* Renamed the support functions associated with active_cdrs_master
appropriately.
ASTERISK-27335
Change-Id: I6104bb3edc3a0b7243ce502e45e8832b0cff14f7
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is because of a n*m loop used when
processing Party B information.
* Added a new CDR container that is keyed to Party B so we don't need such
a large loop when processing Party B information.
NOTE: To reduce the size of the patch I deferred to another patch the
renaming of the Party A active_cdrs_by_channel container to
active_cdrs_master and renaming the container's hash and cmp functions
appropriately.
ASTERISK-27335
Change-Id: I0bf66e8868f8adaa4b5dcf9e682e34951c350249
It's possible for bfdobj to be created but syms not created. If syms
was not allocated in the current loop iteration but was allocated in the
previous iteration it would crash.
ASTERISK-27340
Change-Id: I5b110c609f6dfe91339f782a99a431bca5837363
This avoids a crash on stopping a chan_sip which failed to start its TLS server.
ASTERISK-27339 #close
Change-Id: I327fc70db68eaaca5b50a15c7fd687fde79263d5
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is we were getting the global config
to determine if we needed to log a debugging message.
* Many calls to ao2_global_obj_ref() were just so we could determine if
debug mode is enabled. Made a global flag to check instead.
* Eliminated many RAII_VAR() usages associated with the remaining
ao2_global_obj_ref() calls.
* Added missing NULL checks for the returned ao2_global_obj_ref() value.
ASTERISK-27335
Change-Id: Iceaad93172862f610cad0188956634187bfcc7cd
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is we were getting the global config
even if we didn't need it.
* Most uses of the global config were only needed on off nominal code
paths so it makes sense to not get it until absolutely needed.
ASTERISK-27335
Change-Id: I00c63b7ec233e5bfffd5d976f05568613d3c2365
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is we were repeatedly setting string
fields to potentially the same string in base_process_party_a(). Setting
a string field involves allocating room for the new string out of a memory
pool which may have to allocate even more memory.
* Check to see if the string field is already set to the desired string.
ASTERISK-27335
Change-Id: I3ccb7e23f1488417e08cafe477755033eed65a7c
The string comparisons for setting these CDR variables was inverted. We
were repeatedly setting these CDR variables only if the channel snapshots
had the same value.
ASTERISK-27335
Change-Id: I9482073524411e7ea6c03805b16de200cb1669ea
* Store weak proxy objects in instances container.
* Remove special unreference function and replace with macro that calls
ao2_cleanup.
* Add REF_DEBUG information to ast_sorcery_open.
Change-Id: I5a150a4e13cee319d46b5a4654f95a4623a978f8
This function finds a weak proxy in an ao2_container and returns the
real object associated with it.
Change-Id: I9da822049747275f5961b5c0a7f14e87157d65d8
Copy the list of weakproxy callbacks to temporary memory so they can be
run without holding the weakproxy lock.
Change-Id: Ib167622a8a0f873fd73938f7611b2a5914308047
The only caller of cdr_object_fn_table.process_party_b() explicitly does
the check before calling.
Change-Id: Ib0c53cdf5048227842846e0df9d2c19117c45618
Since ASTERISK-26922, this issue affected only those chan_sip which were
* enabled for dual-stack (bindaddr=::), and
* enabled for TCP (tcpenable=yes) and/or TLS (tlsenable=yes), and
* tried to register and/or invite a IPv4-only service,
* via TCP and/or TLS.
Now, ast_tcptls_client_create does not re-bind to [::] anymore.
ASTERISK-27324 #close
Change-Id: I4b242837bdeb1ec7130dc82505c6180a946fd9b5
ast_strings_match uses sscanf and checks for non-zero return to verify a
token was parsed. This is incorrect as sscanf returns EOF (-1) for errors.
ASTERISK-27318 #close
Change-Id: Ifcece92605f58116eff24c5a0a3b0ee08b3c87b1
Some endpoints do not like a stream being reused for a new
media stream. The frame/jitterbuffer can rely on underlying
attributes of the media stream in order to order the packets.
When a new stream takes its place without any notice the
buffer can get confused and the media ends up getting dropped.
This change uses the SSRC change to determine that a new source
is reusing an existing stream and then bridge_softmix renegotiates
each participant such that they see a new media stream. This
causes the frame/jitterbuffer to start fresh and work as expected.
ASTERISK-27277
Change-Id: I30ccbdba16ca073d7f31e0e59ab778c153afae07
When a sip session is refreshed, the stream topology is looped
through, checking each stream for compatible formats. This would
cause a crash if the stream state was AST_STREAM_STATE_REMOVED,
since the formats would never be set for this stream, causing
a NULL value to be returned from ast_stream_get_formats. This
commit adds a check for streams with removed states.
Also removed a stray semicolon.
Change-Id: Ic86f8b65a4a26a60885b28b8b1a0b22e1b471d42
When two channels were early bridged in a native_rtp bridge, the RTP description
on one side was not updated when the other side answered.
This patch forbids non-answered channels to enter a native_rtp bridge, and
triggers a bridge reconfiguration when an ANSWER frame is received.
ASTERISK-27257
Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df
The Websocket implementation will steal the underlying stream of
TCP/TLS sessions. This results in an error message being output
about a stream not being present when in reality this is actually
fine.
This change moves it to a debug message instead.
Change-Id: I66cc639080b4b4599beadb4faa7d313f2721d094
Jenkins2
Corey Farrell
Alexander Traud
Sean Bright
Joshua Colp
Sungtae Kim
Richard Mudgett
krells
Tzafrir Cohen
George Joseph
Ivan Poddubny
Kevin Harwell
Niklas Larsson
Ben Ford
Thomas Sevestre
Jean Aunis