This adds documentation to handle_cli_malloc_trim() indicating how it
can be useful when debugging OOM conditions.
Change-Id: I1936185e78035bf123cd5e097b793a55eeebdc78
We've had multiple opportunities where Richard Mudgett's
malloc_trim patch has been useful. Let's get it
pushed up to gerrit and merged.
Since malloc_trim is only available in libc, an entry is
added to configure.ac to create a definition for
HAVE_MALLOC_TRIM.
Change-Id: Ia38308c550149d9d6eae4ca414a649957de9700c
When a channel snapshot was created it used to be done
from scratch, copying all data (many strings). This incurs
a cost when doing so.
This change segments the channel snapshot into different
components which can be reused if unchanged from the
previous snapshot creation, reducing the cost. In normal
cases this results in some pointers being copied with
reference count being bumped, some integers being set,
and a string or two copied. The other benefit is that it
is now possible to determine if a channel snapshot update
is redundant and thus stop it before a message is published
to stasis.
The specific segments in the channel snapshot were split up
based on whether they are changed together, how often they
are changed, and their general grouping. In practice only
1 (or 0) of the segments actually get changed in normal
operation.
Invalidation is done by setting a flag on the channel when
the segment source is changed, forcing creation of a new
segment when the channel snapshot is created.
ASTERISK-28119
Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423
Channels no longer use the Stasis cache for channel snapshots. Instead
they are stored in a hash table in stasis_channels which reduces the
number of Stasis messages created and allows better storage.
As a result the following APIs are no longer available since the stasis
cache is no longer used:
ast_channel_topic_cached()
ast_channel_topic_all_cached()
The ast_channel_cache_all() and ast_channel_cache_by_name() functions
now return an ao2_container of ast_channel_snapshots rather than
a container of stasis_messages therefore you can't (and don't need
to) call stasis_cache functions on it.
The ast_channel_topic_all() function now returns a normal topic not
a cached one so you can't use stasis cache functions on it either.
The ast_channel_snapshot_type() stasis message now has the
ast_channel_snapshot_update structure as it's data. It contains the
last snapshot and the new one.
ast_channel_snapshot_get_latest() still returns the latest snapshot.
The latest snapshot is now stored on the channel itself to eliminate
cache hits when Stasis messages that have the snapshot as a payload
are created.
ASTERISK-28102
Change-Id: I9334febff60a82d7c39703e49059fa3a68825786
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
* ast_cli_complete
* ast_complete_channels
* ast_complete_applications
These generators will now use ast_cli_completion_add if state == -1.
Change-Id: I7ff311f0873099be0e43a3dc5415c0cd06d15756
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
* 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
The completion generator is missing a return so typing "core set debug
all off <tab>" causes the command to actually execute.
Change-Id: Ibf6462088a74eee66967732b50445783ebefc20b
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
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
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 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
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
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
GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
ASTERISK-27156 #close
Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
The "core show channel" CLI command will now output the streams
present on the channel with their details.
ASTERISK-26811
Change-Id: I9c95b57aa09415005f0677a1949a0feb07e4987a
* app_minivm: Use built-in completion facilities to complete optional
arguments.
* app_voicemail: Use built-in completion facilities to complete
optional arguments.
* app_confbridge: Add missing colons after 'Usage' text.
* chan_alsa: Use built-in completion facilities to complete optional
arguments.
* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'
* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'
* func_odbc: Correct completions for 'odbc read' and 'odbc write'
* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.
* main/bridge: Correct completions for 'bridge kick.'
* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.
* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'
* main/pbx_app: Remove redundant completions for 'core show
applications.'
* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'
* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.
Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes
all traces of it.
Previously exported symbols removed:
* __ast_register_file
* __ast_unregister_file
* ast_complete_source_filename
This also removes the mtx_prof static variable that was declared when
MTX_PROFILE was enabled. This variable was only used in lock.c so it
is now initialized in that file only.
ASTERISK-26480 #close
Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
Since Asterisk 1.8, the command "core set debug" on the command-line interface
asks not for a file (.c) but a module name. This change shows modules (.so) on
the auto-completion via a tabulator or the question mark. Now, when you
partially type a module name, TAB or ?, you get the correct candidiates.
ASTERISK-26480
Change-Id: I1213f1dd409bd4ff8de08ad80cb0c73cafb1bae0
POSIX defines signal.h. sys/signal.h should not be used as it is
c-library internal header which may or may not exist. Notably with
musl it generates warning of being incorrect.
Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc
During stress testing, we have frequently seen crashes occur because a
CLI or AMI command attempts to access information that is in the process
of being destroyed.
When addressing how to fix this issue, we initially considered fixing
individual crashes we observed. However, the changes required to fix
those problems would introduce considerable overhead to the nominal
case. This is not reasonable in order to prevent a crash from occurring
while Asterisk is already shutting down.
Instead, this change makes it so AMI and CLI commands cannot be executed
if Asterisk is being shut down. For AMI, this is absolute. For CLI,
though, certain commands can be registered so that they may be run
during Asterisk shutdown.
ASTERISK-25825 #close
Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990
Refactor and created function ast_cli_print_timestr_fromseconds to print
seconds formatted: year(s) week(s) day(s) hour(s) second(s)
This function now is used in addons/cdr_mysql.c,cdr_pgsql.c, main/cli.c,
res_config_ldap.c, res_config_pgsql.c.
Change-Id: Ibeb8634102cd11d3f8623398b279cb731bcde36c
Because the context, extension, and application are stored in stringfields,
checking for them being NULL doesn't work so well. This patch uses the
appropriate string library call, ast_strlen_zero, to see if there is a value
in the context/exten/app values.
Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23
* Pass module to ast_cli_register and ast_cli_register_multiple.
* Add a module reference before executing any CLI callback, remove
the reference when complete.
ASTERISK-25049 #close
Reported by: Corey Farrell
Change-Id: I7aafc7c9f2b912918f28fe51d51e9e8a755750e3
This change modifies how the the output from a CLI command is sent
to a client over AMI.
Output from the CLI command is now sent as a series of zero-or-more
Output: headers.
Additionally, commands that fail to execute (eg: no such command,
invalid syntax etc.) now cause an Error response instead of Success.
If the command executed successfully, but the manager unable to
provide the output the reason will be included in the Message:
header. Otherwise it will contain 'Command output follows'.
Depends on a new version of starpy (> 1.0.2) that supports the new
output format.
See pull-request https://github.com/asterisk/starpy/pull/34
ASTERISK-24730
Change-Id: I6718d95490f0a6b3f171c1a5cdad9207f9a44888
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
remove passing the version in with the macro. Other facilities
than 'core show file version' make use of the file names, such as
setting a debug level only on a specific file. As such, the act of
registering source files with the Asterisk core still has use. The
macro rename now reflects the new macro purpose.
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Remove the "core show file version" CLI command. Without the file
version, it is no longer useful.
- Remove the ast_file_version_find function. The file version is no
longer tracked.
- Rename ast_register_file_version/ast_unregister_file_version to
ast_register_file/ast_unregister_file, respectively.
* main/manager: Remove value from the Version key of the ModuleCheck
Action. The actual key itself has not been removed, as doing so would
absolutely constitute a backwards incompatible change. However, since
the file version is no longer tracked, there is no need to attempt to
include it in the Version key.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action
- Removal of the "core show file version" CLI command
Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups. Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe. ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.
Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.
ASTERISK-24142 #close
Reported by: David Brillert
ASTERISK-24683 #close
Reported by: Peter Katzmann
ASTERISK-24805 #close
Reported by: Badalian Vyacheslav
ASTERISK-24881 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
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Switch logger callid's from AO2 objects to simple integers.
This helps in two ways. Copying integers is faster than
referencing AO2 objects, so this will result in a small
reduction in logger overhead. This also erases the possibility
of an infinate loop caused by an invalid callid in
threadstorage.
ASTERISK-24833 #comment Committed callid conversion to trunk.
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4466/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
During some performance testing of Asterisk with AGI, ARI, and lots of Local
channels, we noticed that there's quite a hit in performance during channel
creation and releasing to the dialplan (ARI continue). After investigating
the performance spike that occurs during channel creation, we discovered
that we create a lot of channel snapshots that are technically unnecessary.
This includes creating snapshots during:
* AGI execution
* Returning objects for ARI commands
* During some Local channel operations
* During some dialling operations
* During variable setting
* During some bridging operations
And more.
This patch does the following:
- It removes a number of fields from channel snapshots. These fields were
rarely used, were expensive to have on the snapshot, and hurt performance.
This included formats, translation paths, Log Call ID, callgroup, pickup
group, and all channel variables. As a result, AMI Status,
"core show channel", "core show channelvar", and "pjsip show channel" were
modified to either hit the live channel or not show certain pieces of data.
While this is unfortunate, the performance gain from this patch is worth
the loss in behaviour.
- It adds a mechanism to publish a cached snapshot + blob. A large number of
publications were changed to use this, including:
- During Dial begin
- During Variable assignment (if no AMI variables are emitted - if AMI
variables are set, we have to make snapshots when a variable is changed)
- During channel pickup
- When a channel is put on hold/unhold
- When a DTMF digit is begun/ended
- When creating a bridge snapshot
- When an AOC event is raised
- During Local channel optimization/Local bridging
- When endpoint snapshots are generated
- All AGI events
- All ARI responses that return a channel
- Events in the AgentPool, MeetMe, and some in Queue
- Additionally, some extraneous channel snapshots were being made that were
unnecessary. These were removed.
- The result of ast_hashtab_hash_string is now cached in stasis_cache. This
reduces a large number of calls to ast_hashtab_hash_string, which reduced
the amount of time spent in this function in gprof by around 50%.
#ASTERISK-23811 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3568/
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The per console verbose level feature as previously implemented caused a
large performance penalty. The fix required some minor incompatibilities
if the new rasterisk is used to connect to an earlier version. If the new
rasterisk connects to an older Asterisk version then the root console
verbose level is always affected by the "core set verbose" command of the
remote console even though it may appear to only affect the current
console. If an older version of rasterisk connects to the new version
then the "core set verbose" command will have no effect.
* Fixed the verbose performance by not generating a verbose message if
nothing is going to use it and then filtered any generated verbose
messages before actually sending them to the remote consoles.
* Split the "core set debug" and "core set verbose" CLI commands to remove
the per module verbose support that cannot work with the per console
verbose level.
* Added a silent option to the "core set verbose" command.
* Fixed "core set debug off" tab completion.
* Made "core show settings" list the current console verbosity in addition
to the root console verbosity.
* Changed the default verbose level of the 'verbose' setting in the
logger.conf [logfiles] section. The default is now to once again follow
the current root console level. As a result, using the AMI Command action
with "core set verbose" could again set the root console verbose level and
affect the verbose level logged.
(closes issue AST-1252)
Reported by: Guenther Kelleter
Review: https://reviewboard.asterisk.org/r/3114/
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A common idiom in Asterisk is to due something like:
for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) {
...do stuff...
}
This is nice because it automatically takes care of the object references
for you. However, there is a pitfall here. If a break statement is in the
for loop, then the current reference is not cleaned up. In some cases, this
is on purpose, but in others there is a leak. This commit fixes the leak
cases.
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The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.
This patch renames the variables, adding the ast_ prefix so they will
be exported.
Review: https://reviewboard.asterisk.org/r/2737
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.
To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.
In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:
single_topic ----------------> all_topic
^
|
single_topic_cached ----+----> all_topic_cached
|
+----> cache
This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.
Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.
(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3