We've been seeing crashes in libbfd when we attempt to generate
a stack trace from multiple threads. It turns out that libbfd
is NOT thread-safe. It can cache the bfd structure and give it to
multiple threads without protecting itself. To get around this,
we've added a global mutex around the bfd functions and also have
refactored the use of those functions to be more efficient and
to provide more information about inlined functions.
Also added a few more tests to test_pbx.c. One just calls
ast_assert() and the other calls ast_log_backtrace(). Neither are
run by default.
WARNING: This change necessitated changing the return value of
ast_bt_get_symbols() from an array of strings to a VECTOR of
strings. However, the use of this function outside Asterisk is not
likely.
ASTERISK-28140
Change-Id: I79d02862ddaa2423a0809caa4b3b85c128131621
When a subscribe or unsubscribe occurs a message is published
containing this information. This change makes it so that the
message no longer uses stringfields or a lock, as both are not
really needed for the message.
Change-Id: I3f4831931d79f94fd979baf48048738df5dc1632
This change adds the ability for subscriptions to indicate
which message types they are interested in accepting. By
doing so the filtering is done before being dispatched
to the subscriber, reducing the amount of work that has
to be done.
This is optional and if a subscriber does not add
message types they wish to accept and set the subscription
to selective filtering the previous behavior is preserved
and they receive all messages.
There is also the ability to explicitly force the reception
of all messages for cases such as AMI or ARI where a large
number of messages are expected that are then generically
converted into a different format.
ASTERISK-28103
Change-Id: I99bee23895baa0a117985d51683f7963b77aa190
Task processors are retrieved using a 'get or create' pattern. The
singleton container was unlocked between the get and create steps so
it's possible that two threads could create task processors with the
same name at the same time.
Change-Id: Id64fae94a6a1e940ddf38fde622dcd4391635382
We cannot use need_el_end and SIGURG when restarting. Instead we need
to run el_end within the SIGHUP restartnow handler.
ASTERISK-28158
Change-Id: Ia852276363c81bdcf1aa29eb4558c5c2fa1218a0
When dn_expand was being called on SRV and NAPTR results, the
return value was being used to calculate the size of the buffer
needed to store the host names. Since dn_expand returns the
length of the COMPRESSED name the buffer could be too short
to hold the EXPANDED name. The expanded name is NULL terminated
so using strlen() is the correct way to determine the length
actually needed for the buffer.
ASTERISK-28127
Reported by: Jan Hoffmann
patches:
patch.diff submitted by janhoffmann (license 6986)
Change-Id: I4d35d6c431c6c6836cb61d37b1378cc47f0b414d
Merge storage for the stats object and name string into the main
allocation for struct ast_taskprocessor.
Change-Id: I74fe9a7f357f0e6d63152f163cf5eef6428218e1
* Ignore console=yes configuration option in remote console processes.
* Use new flag to tell consolethread to run el_end and exit when needed.
ASTERISK-28158
Change-Id: I9e23b31d4211417ddc88c6bbfd83ea4c9f3e5438
Default logging was not setup correctly when there was no logger.conf.
This resulted in many expected log messages not actually getting out to
the console.
Change-Id: I542e61c03b2f630ff5327f9de5641d776c6fa70c
Adding the "label" attribute used for participant info correlation
was previously done in app_confbridge but it wasn't working
correctly because it didn't have knowledge about which video
streams belonged to which channel. Only bridge_softmix has that
data so now it's set when the bridge topology is changed.
ASTERISK-28107
Change-Id: Ieddeca5799d710cad083af3fcc3e677fa2a2a499
These macros have been documented as legacy for a long time but are
still used in new code because they exist. Remove all references to:
* ao2_container_alloc_options
* ao2_t_container_alloc_options
* ao2_t_container_alloc
These macro's are also removed. Only ao2_container_alloc remains due to
it's use in over 100 places.
Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a
__ast_mutex_logger used the variable `canlog` without accepting it as a
argument. Replace with internal macro `log_mutex_error` which takes
canlog as the first arguement. This will prevent confusion when working
with lock.c code, many of the function declare the canlog variable and
in some cases it previously appeared to be unused.
Change-Id: I83b372cb0654c5c18eadc512f65a57fa6c2e9853
Most were in comments. A couple were in warning messages.
Pointed out by Jonathan H on the Asterisk users mailing list.
Change-Id: I6286939dff5d0a27a2758140570106f1cb351855
Add attribute_warn_unused_result to ast_taskprocessor_push,
ast_taskprocessor_push_local and ast_threadpool_push. This will help
ensure we perform the necessary cleanup upon failure.
Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d
This allows us to process AO2 statistics for total objects, memory
usage, memory overhead and lock usage.
* Install refstats.py and reflocks.py into the Asterisk scripts folder.
* Enable support for reflocks.py without DEBUG_THREADS.
Steal a bit from the ao2 magic to flag when an object lock is used.
Remove 'lockobj' from reflocks.py since we can now record 'used' or
'unused' for those objects.
Add comments to explain thread safety of the 'struct __priv_data'
bitfields.
Change-Id: I84e9d679cc86d772cc97c888d9d856a17e0d3a4a
thread_worker_pair, set_size_data and task_pushed_data structures are
allocated with AO2 objects, passed to a taskprocessor, then released.
They never have multiple owners or use locking so AO2 only adds
overhead.
Change-Id: I2204d2615d9d952670fcb48e0a9c0dd1a6ba5036
The field used to store call arguments was not large enough to hold the
arguments string that can be constructed for 'open'. Expand it to
prevent this warning/error.
Change-Id: I514927f256481bc84df10a51b19d5b5fb1bc387e
ast_sendtext_data() would create an incorrect T.140 text frame which
length include the null terminator byte. It causes ultimately RTP
packets to be send with this trailing 0. The proposed fix just set the
correct length to the text frame
ASTERISK-28089
Reported by: Emmanuel BUU
Tested by: Emmanuel BUU
Change-Id: I7ab1b9ed1e21683b2b667ea0a59d9aba3c77dd96
This has no effect on startup since AST_MODULE_LOAD_FAILURE aborts
startup, but it's possible for this code to be returned on manual load
of a module after startup.
It is an error for a module to not have a load callback but this is not
a fatal system error. In this case flag the module as declined, return
AST_MODULE_LOAD_FAILURE only if a required module is broken.
Expand doxygen documentation for AST_MODULE_LOAD_*.
Change-Id: I3c030bb917f6e5a0dfd9d91491a4661b348cabf8
There is currently no way to indicate to Asterisk that TLS certificates
and/or keys have been updated other than by modifying http.conf or
restarting Asterisk.
There is already code in main/tcptls.c that determines if a reload is
actually necessary based on the hashes of the certicate and dependent
files, so this change merely gives us a way to request a reload without
explicitly modifying http.conf.
Change-Id: Ie795420dcc7eb3d91336820688a29adbcc321276
* ACO options
* Indications
* Module loader ref_debug object
* Media index info and variants
* xmldoc items
These allocation locations were identified using reflocks.py on the
master branch.
Change-Id: Ie999b9941760be3d1946cdb6e30cb85fd97504d8
When a module reload fails we never set AST_MODULE_RELOAD_ERROR. This
caused reload failures to incorrectly report 'No module found'.
Change-Id: I5f3953e0f7d135e53ec797f24c97ee3f73f232e7
* Display list of unavailable dependencies when they cause another
module to fail loading.
* When a module declines to load find all modules which depend on it so
they can be declined and listed together.
* Prevent retry of declined modules during startup.
* When a module fails to dlopen try loading it with RTLD_LAZY so we can
attempt to display the list of missing dependencies.
These changes are meant to reduce logger spam that is caused when a
module has many dependencies and declines to load. This also fixes some
error paths which failed to recognize required modules.
Module load/start errors are delayed until the end of loader startup.
Change-Id: I046052c71331c556c09d39f47a3b92975f3e1758
If a channel creates an AST_TEXT_FRAME with datalen == 0, the ast_frdup()
and ast_frisolate() functions could create a clone frame with an invalid
data.ptr which would cause a crash. The proposed fix is to make sure that
for such empty text frames, ast_frdup() and ast_frisolate() return cloned
text frames with a valid data.ptr.
ASTERISK-28076
Reported by: Emmanuel BUU
Tested by: Emmanuel BUU
Change-Id: Ib882dd028598f13c4c233edbfdd7e54ad44a68e9
When DEBUG_THREADS is enabled we can know if the astobj2 mutex / rwlock
was ever used, so it can be recorded in the REF_DEBUG destructor entry.
Create contrib/scripts/reflocks.py to process locking used by
allocator. This can be used to identify places where
AO2_ALLOC_OPT_LOCK_NOLOCK should be used to reduce memory usage.
Change-Id: I2e3cd23336a97df2692b545f548fd79b14b53bf4
Add a volatile flag to lock tracking structures so we only need to use
the global lock when first initializing tracking.
Additionally add support for DEBUG_THREADS_LOOSE_ABI. This is used by
astobj2.c to eliminate storage for tracking fields when DEBUG_THREADS is
not defined.
Change-Id: Iabd650908901843e9fff47ef1c539f0e1b8cb13b
Reduce options to 2-bit field, magic to 30 bit field. Move ref_counter
next to options and explicitly use int32_t so the fields will pack.
This reduces memory overhead for every ao2 object by 8 bytes on x86_64.
Change-Id: Idc1baabb35ec3b3d8de463c4fa3011eaf7fcafb5
* In main/config.c, AST_INCLUDE_GLOB is fixed to '1' making the #ifdefs
pointless.
* In utils/extconf.c, AST_INCLUDE_GLOB is never defined so there is a
lot of dead code.
Change-Id: I1bad1a46d7466ddf90d52cc724e997195495226c
When REF_DEBUG and AO2_DEBUG are both enabled we closed the refs log
before we shutdown astobj2_container. This caused the AO2_DEBUG
container registration container to be reported as a leak.
Change-Id: If9111c4c21c68064b22c546d5d7a41fac430430e
Use json_vsprintf from versions which contain fix for va_copy leak.
Apply fixes from jansson master:
* va_copy leak fix.
* Avoid potential invalid memory read in json_pack.
* Rename variable that shadowed another.
Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539
* Use "o*" format specifier for optional fields in ast_json_party_id.
* Stop using ast_json_deep_copy on immutable objects, it is now thread
safe to just use ast_json_ref.
Additional changes to ast_json_pack calls in the vicinity:
* Use "O" when an object needs to be bumped. This was previously
avoided as it was not thread safe.
* Use "o?" and "O?" to replace NULL with ast_json_null(). The
"?" is a new feature of ast_json_pack starting with Asterisk 16.
Change-Id: I8382d28d7d83ee0ce13334e51ae45dbc0bdaef48
app_voicemail wasn't properly cleaning up the stasis cache or the
mwi topic pool when the module was unloaded or when a user was
deleted as a result of a reload. This resulted in leaks in both
areas.
* app_voicemail now calls ast_delete_mwi_state_full when it frees
a user structure and ast_delete_mwi_state_full in turn now calls
the new stasis_topic_pool_delete_topic function to clear the topic
from the pool.
Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8
When writing an RTCP report to json the code attempts to pack the "ssrc" and
"source_ssrc" unsigned integer values as a signed int value type. This of course
means if the ssrc's unsigned value is greater than that which can fit into a
signed integer value it gets converted to a negative number. Subsequently, the
negative value goes out in the json report.
This patch now packs the value as a json_int_t, which is the widest integer type
available on a given system. This should make it so the value no longer
overflows.
Note, this was caught by two failing tests hep/rtcp-receiver/ and
hep/rtcp-sender.
Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0
does_id_conflict() was passing a pointer to an int to a callback
that expected a pointer to a size_t.
Found by the Address Sanitizer.
Change-Id: I0ff542067eef63a14a60301654d65d34fe2ad503
There's been a long standing leak when using topic pools. The
topics in the pool get cleaned up when the last pool reference is
released but you can't remove a topic specifically. If you reloaded
app_voicemail for instance, and mailboxes went away, their topics
were left in the pool.
* Added stasis_topic_pool_delete_topic() so modules can clean up
topics from pools.
* Registered the topic pool containers so it can be examined from
the CLI when AO2_DEBUG is enabled. They'll be named
"<topic_pool_name>-pool".
Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25
Stasis message types are global ao2 objects and we make stasis messages
and cache entries hold references to them. Since there are currently
situations where cache objects are never deleted, the reference count on
the types can exceed 100000 and generate a FRACK assertion message. The
stasis message cache could conceivably also have that many messages
legitimately on large systems.
The only down side to not holding the message type ref in the stasis
message is it only makes a crash either at shutdown or when manually
unloading a busy module slightly more likely. However, this is more
exposing a pre-existing stasis shutdown ordering issue than a problem with
not holding a message type ref in stasis messages.
* Made stasis messages and cache entries no longer hold a ref to the
message type.
Change-Id: Ibaa28efa8d8ad3836f0c65957192424c7f561707
* Create the stasis message object without a lock as it is immutable.
* Create the stasis message type object without a lock as it is immutable.
* Creating the stasis message type could crash if the passed in type name
is NULL and REF_DEBUG is enabled. Added missing NULL check when passing
the ao2 object tag string.
Change-Id: I28763c58bb9f0b427c11971d0103bf94055e7b32
Since app_voicemail no longer uses the cache to maintain its state
there is no longer a need to cache these messages.
ASTERISK-27121
Change-Id: I321c708505f5ad8d00e1b0afc4c27dc2ac12ecb4
The documentation already specified EVENT_FLAG_DIALPLAN for this
event, but the implementation was using EVENT_FLAG_CALL.
Using EVENT_FLAG_DIALPLAN allows AMI clients to opt out of receiving
this highly verbose event.
ASTERISK-28033
Change-Id: I45b3119f30e4dbc17b49831f2b1a4f2c1beadafe
Generators provide such functionality as tone generation or
silence generation. RTCP frames provide RTCP information and
should not stop generators from operating.
ASTERISK-28005
Change-Id: Ieadada07b068a7aa426e8763f1b73a18e1ac34a9
The stasis cache provides a way to reconstruct the current state
of topic subscribers. Unfortunately, since every subscribe and
unsubscribe is cached, the cache continues to grow unabated while
asterisk is running. This patch removes subscribe messages from
the cache when the corresponding unsubscribe is received.
This patch also registers the cache containers with ao2 so that if
AO2_DEBUG is turned on, you can list the container and get its
stats from the CLI.
ASTERISK-27121
Change-Id: I3d18905e477f3721815da91f30da8d3fbb2d4f56
Added a check when we receive a HTTP request line or header line that is
too long. We now return an error response to the sender because we are
not able to process the request.
Change-Id: I6df2705435fd7dde4d5d3bdf7acec859cfb7c12d
Providing a buffer larger than the internal buffer of ast_iostream_gets()
fails to get lines longer than the internal buffer.
* Made ast_iostream_gets() fill the supplied buffer with read data until
either a '\n' is found or the supplied buffer is filled just like fgets().
Change-Id: If18b3f6ee500e22f0633a68779ed09f7e0f305ed
All HTTP/AMI message headers are being sent to the verbose channel.
There are multiple places this is happening. Consolidate the loop into
a function. Drop the debug/verbose message.
Convert to using ast_asprintf to perform the length calculation, memory
allocation and snprintf all in one step.
Change-Id: Ic45e673fde05bd544be95ad5cdbc69518207c1a1
When the stasis cache is used a hash is calculated for
retrieving or inserting messages. This change calculates
a hash when the message type is initialized that is then
used each time needed. This ensures that the hash is
calculated only once for the message type.
Change-Id: I4fe6bfdafb55bf5c322dd313fbd8c32cce73ef37
Changing any Menuselect option in the `Compiler Flags` section causes a
full rebuild of the Asterisk source tree. Every enabled option causes
a #define to be added to buildopts.h, thus breaking ccache caching for
every source file that includes "asterisk.h". In most cases each option
only applies to one or two files. Now we only define those options for
the specific sources which use them, this causes much better cache
matching when working with multiple builds. For example testing code
with an without MALLOC_DEBUG will now use just over half the ccache
size, only main/astmm.o will have two builds cached instead of every
file.
Reorder main/Makefile so _ASTCFLAGS set on specific object files are all
together, sorted by filename. Stop adding -DMALLOC_DEBUG to CFLAGS of
bundled pjproject, this define is no longer used by any header so only
serves to break cache.
The only code change is a slight adjustment to how main/astmm.c is
initialized. Initialization functions always exist so main/asterisk.c
can call them unconditionally. Additionally rename the astmm
initialization functions so they are not exported.
Change-Id: Ie2085237a964f6e1e6fff55ed046e2afff83c027
With the new module loader it was missed that built-in modules never
parsed dependencies from mod->info into vectors of mod. This caused
manager to be initialized before acl (named_acl). If manager.conf
used any named ACL's they would not be found and result in no ACL being
applied to the AMI user.
In addition to the manager ACL fix this adds "extconfig" to all builtin
modules which support realtime configuration. This only matters if one
of the builtin modules is configured with 'preload', depending on
"extconfig" will cause config.c to automatically be initialize during
the preload stage.
Change-Id: I482ed6bca6c1064b05bb538d7861cd7a4f02d9fc
When converting from a json object to an ast variables list the conversion
algorithm was doing a complete traversal of the entire variables list for
every item appended from the json structure.
This patch makes it so the list is no longer traversed for each new ast
variable being appended.
Change-Id: I8bf496a1fc449485150d6db36bfc0354934a3977
When publishing a device state the change can be marked as being
cachable or not. If it is not cached the change is just published
to all interested and not stored away for later query. This was not
fully taken into account when publishing in stasis. The act of
publishing would create a topic for the device even if it may be
ephemeral.
This change makes it so messages which are not cached won't create
a topic for the device. If a topic does already exist it will be
published to but otherwise the change will only be published to
the device state all topic.
ASTERISK-27591
Change-Id: I18da0e8cbb18e79602e731020c46ba4101e59f0a
The "xmldoc dump" cli command was simply concatenating xml documents
into the output file. The resulting file had multiple "xml"
processing instructions and multiple root elements which is illegal.
Normally this isn't an issue because Asterisk has only 1 main xml
documentation file but codec_opus has its own file so if it's
downloaded and you do "xmldoc dump", the result is invalid.
* Added 2 new functions to xml.c:
ast_xml_copy_node_list creates a copy of a list of children.
ast_xml_add_child_list adds a list to an existing list.
* Modified handle_dump_docs to create a new output document and
add to it the children from each input file. It then dumps the
new document to the output file.
Change-Id: I3f182d38c75776aee76413dadd2d489d54a85c07
In the past there was an assertion in the ast_sched_del function
and in order to ensure it was useful the calling function name,
line number, and filename had to be passed in. This cause the ABI
to be different between dev mode and non-dev mode.
This assertion is no longer present so the special logic can be
removed to make it the same between them both.
Change-Id: Icbc69c801e357d7004efc5cf2ab936d9b83b6ab8
Support has been added for receiving a NACK request and handling it.
Now, Asterisk can detect when a NACK request should be sent and knows
how to construct one based on the packets we've received from the remote
end. A buffer has been added that will store out of order packets until
we receive the packet we are expecting. Then, these packets are handled
like normal and frames are queued to the core like normal. Asterisk
knows which packets to request in the NACK request using a vector
which stores the sequence numbers of the packets we are currently missing.
If a missing packet is received, cycle through the buffer until we reach
another packet we have not received yet. If the buffer reaches a certain
size, send a NACK request. If the buffer reaches its max size, queue all
frames to the core and wipe the buffer and vector.
According to RFC3711, the NACK request must be sent out in a compound
packet. All compound packets must start with a sender or receiver
report, so some work was done to refactor the current sender / receiver
code to allow it to be used without having to also include sdes
information and automatically send the report.
Also added additional functionality to ast_data_buffer, along with some
testing.
For more information, refer to the wiki page:
https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements
ASTERISK-27810 #close
Change-Id: Idab644b08a1593659c92cda64132ccc203fe991d
I have removed the STATIC_BUILD option immediately as it has not
been maintained in many years and is non-functional.
ASTERISK-27965
Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7
Fixes issue where error msg
"Use of before/init after destruction"
was being printed on disabled messages
in dev mode. With this
fix if message is disabled
a warning will print.
ASTERISK-25548
Change-Id: Ie0d866d1cbc60c16dbef08bc65e99505c3c1adfa
* Merge the preload and load stages, use load ordering to try preload's
first. This fixes an issue where `preload=res_config_curl` would fail
unless res_curl and func_curl were also preloaded. Now it is only
required that those modules be loaded during startup: autoload or
regular load is good enough.
* The configuration option `require` and `preload-require` were only
effective if the modules failed to load. These options will now abort
Asterisk startup if required modules fail to reach the 'Running'
state.
* Missing or invalid 'module.conf' did not prevent startup. Asterisk
doesn't do anything without modules so this a fatal error.
Change-Id: Ie4176699133f0e3a823b43f90c3348677e43a5f3
Separate "name" into "classname" and "name".
Use '.' for classname separator instead of '/'.
Prefix reserved words with '_'.
Wrap output with a top-level "testsuites" element.
Change-Id: Iec1a985eba1c478e5c1d65d5dfd95cb708442099
There is a rare case (do to the infrequent timing involved) where
CDR submission threads in batch mode can deadlock with a currently
running CDR batch process. This patch should remove the need for
holding the lock in the scheduler and should clean a few code
paths up that inconsistently submitted new work to the CDR batch
processor.
ASTERISK-27909
Change-Id: I6333e865db7c593c102c2fd948cecdb96481974d
Reported-by: Denis Lebedev
The AMI action was directly sending the text to the channel driver.
However, this makes two threads attempt to handle media and runs afowl of
CHECK_BLOCKING.
* Queue a read action to make the channel's media handling thread actually
send the text message. This changes the AMI actions success/fail response
to just mean the text was queued to be sent not that the text actually got
sent. The channel driver may not even support sending text messages.
ASTERISK-27943
Change-Id: I9dce343d8fa634ba5a416a1326d8a6340f98c379
In addition to text/* content types, incoming_in_dialog_request now
accepts application/* content types.
Also fixed a length issue when copying the body text. It was one
character short.
ASTERISK-27942
Change-Id: I4e54d8cc6158dc47eb8fdd6ba0108c6fd53f2818
* changes:
channel.c: Make CHECK_BLOCKING() save thread LWP id for messages.
channel.c: Fix usage of CHECK_BLOCKING()
autoservice: Don't start channel autoservice if the thread is a user interface.
With ./configure --enable-dev-mode[=noisy], the build fails because every
warning gets an error. Therefore, Asterisk has to be free of warnings and this
variable must go.
Change-Id: I63dd2bc4833b9bdb04602f83422d16caf289d46a
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
ASTERISK-27625
Change-Id: I4d6a2fe7386ea447ee199003bf8ad681cb30454e
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
ASTERISK-27625
Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905
The CHECK_BLOCKING() macro is used to indicate if a channel's handling
thread is about to do a blocking operation (poll, read, or write) of
media. A few operations such as ast_queue_frame(), soft hangup, and
masquerades use the indication to wake up the blocked thread to reevaluate
what is going on.
ASTERISK-27625
Change-Id: I4dfc33e01e60627d962efa29d0a4244cf151a84d
Executing dialplan functions from either AMI or ARI by getting a variable
could place the channel into autoservice. However, these user interface
threads do not handle the channel's media so we wind up with two threads
attempting to handle the media.
There can be one and only one thread handling a channel's media at a time.
Otherwise, we don't know which thread is going to handle the media frames.
ASTERISK-27625
Change-Id: If2dc94ce15ddabf923ed1e2a65ea0ef56e013e49
In Asterisk there are some dynamic codecs that have
a fixed payload number. This number was being improperly
used to negotiate the codec, instead of using the name
and sample rate. This could result in the wrong payload
number being negotiated for a codec.
This change makes it so that only static payloads
will be negotiated using their payload number.
ASTERISK-27848
Change-Id: Ia865830170fd3f808cdb33104f3d4c4ffdc77570
Before Asterisk sends an HTTP response (at least in the case of errors),
it attempts to read & discard the content of the request. If the client
lies about the Content-Length, or the connection is closed from the
client side before "Content-Length" bytes are sent, the request handling
thread will busy loop.
ASTERISK-27807
Change-Id: I945c5fc888ed92be625b8c35039fc6d2aa89c762
ast_indicate is invoked with the bridge locked. As ast_indicate locks the
other end of the bridge as well this can lead to a deadlock in some situations.
(Especially when a different thread does the same in the reverse order).
This patch calls ast_indicate after unlocking the bridge which fixes the
deadlock. Calling ast_indicate with these parameters without locking the
bridge should be safe as this is done at different places without a
bridge lock.
ASTERISK-27094 #close
Reported-by: David Brillert
Change-Id: I5f86c1e2ce75b9929a36ab589b18c450e62ea35f
There was no real reason to limit the conteny type to text/plain other
than that's what it was limited to before. Now any text/* content
type will be allowed for channel drivers that don't support enhanced
messaging and any type will be allowed for channel drivers that do
support enhanced messaging.
Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9
asterisk/tcptls.h was included (explicitly, implicitly, or transitively). Those
inclusions got replaced by forward declarations. As side effect, the inclusions
got completed.
ASTERISK-27878
Change-Id: I9d102728e30336d6522e5e4ae9e964013a0835f7
Additionally, this change allows auto-negotiation of the elliptic curve/group
for servers, not only with OpenSSL 1.0.2 but also with OpenSSL 1.1.0 and newer.
This enables X25519 (since OpenSSL 1.1.0) and X448 (since OpenSSL 1.1.1) as a
side-effect.
ASTERISK-27876
Change-Id: I62c2aba4a630aefc231b71f646207e8c027d9497
When RTP was originally created it had the ability to place a single
extension in an RTP packet. In practice people wanted to potentially
put multiple extensions in one and so RFC 5285 (obsoleted by RFC
8285) came into existence. This allows RTP extensions to be negotiated
with a unique identifier to be used in the RTP packet, allowing
multiple extensions to be present in the packet.
This change extends the RTP engine API to add support for this. A
user of it can enable extensions and the API provides the ability to
retrieve the information (to construct SDP for example) and to provide
negotiated information (from SDP). The end result is that the RTP
engine can then query to see if the extension has been negotiated and
what unique identifier is to be used. It is then up to the RTP engine
implementation to construct the packet appropriately.
The first extension to use this support is abs-send-time which is
defined in the REMB draft[1] and is a second timestamp placed in an
RTP packet which is for when the packet has left the sending system.
It is used to more accurately determine the available bandwidth.
ASTERISK-27831
[1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03
Change-Id: I508deac557867b1e27fc7339be890c8018171588
__ast_channel_alloc_ap() had a failure exit path that hadn't setup the fd
descriptors to -1 yet. The destructor would then attempt to close these
fd's that had never been opened.
Change-Id: Icf21093f36c60781e8cf6ee9d586536302af33e3
The config engine destroy_func callback function returns the number of
rows deleted or -1 on error. But the function
ast_destroy_realtime_fields treated non-zero return values as error.
ASTERISK-27863
Change-Id: Ied02b38e8196cb03043e609a0679feebd288d17b
This function originally was used in chan_sip to enable some simplifying
assumptions and eventually was copy and pasted into res_pjsip_logger and
res_hep. Since it's replicated in three places, it's probably best to
move it into the public netsock2 API for these modules to use.
Change-Id: Id52e23be885601c51d70259f62de1a5e59d38d04
Mantis-3709 (Commit 68ff3c3, Asterisk 1.2) added support for the video format
H.263+. For this, the RTP payload ID 103 got assigned statically. Commit f1aadc8
assigned another payload ID 98 for this format in Asterisk 1.6.
Change-Id: I90e35b158487f8f1f8187da6241b54cd3b74e667
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
* 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
Joshua Colp
George Joseph
Corey Farrell
Richard Mudgett
Sean Bright
Jenkins2
neutrino88
Kevin Harwell
lvl
Jaco Kroon
Ben Ford
Chris-Savinovich
Matthew Fredrickson
Alexander Traud
Pirmin Walthert
Alexei Gradinari
Gaurav Khurana
Florian Floimair
Guido Falsi
Sungtae Kim
krells
Tzafrir Cohen
Ivan Poddubny
Niklas Larsson