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* 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/asterisk: Correct and extend completions for 'core show file
version.'
* 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
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The mechanism used for detecting the maximum log level compiled into the
linked pjproject did not work. The API call simply stores the requested
level into an integer and does no range checking. Asterisk was assuming
that there was range checking and limited the new value to the allowable
range. To get the actual maximum log level compiled into the linked
pjproject we need to get and save off the initial set log level from
pjproject. This is the maximum log level supported.
* Get and save off the initial log level setting before altering it to the
desired level on startup. This has to be done by a macro rather than
calling a core function to avoid incorrectly linking pjproject.
* Split the initial log level warning messages to warn if the linked
pjproject cannot support the requested startup level and if it is too low
to get the pjproject buildopts for "pjproject show buildopts".
* Adjust the CLI "pjproject set log level" to check the saved max log
level and to generate normal output messages instead of a warning message.
ASTERISK-26743 #close
Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4
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Use of the new logging is as simple as issuing the new CLI command or
setting the new pjproject.conf option.
Other options that can affect the logging are how you have the pjproject
log levels mapped to Asterisk log types in pjproject.conf and if you have
configured Asterisk to log the DEBUG type messages. Altering the
pjproject.conf level mapping shouldn't be necessary for most installations
as the default mapping is sensible. Configuring Asterisk to log the DEBUG
message type is standard practice for collecting debug information.
* Added CLI "pjproject set log level" command to dynamically adjust the
maximum pjproject log message level.
* Added CLI "pjproject show log level" command to see the currently set
maximum pjproject log message level.
* Added pjproject.conf startup section "log_level" option to set the
initial maximum pjproject log message level so all messages could be
captured from initialization.
* Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into
bundled pjproject. Pjproject will use the currently set run time log
level to determine if a log message is generated just like Asterisk
verbose and debug logging levels.
* In log_forwarder(), made always log enabled and mapped pjproject log
messages. DEBUG mapped log messages are no longer gated by the current
Asterisk debug logging level.
* Removed RAII_VAR() from res_pjproject.c:get_log_level().
ASTERISK-26630 #close
Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
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If a TCP/TLS connection was pending (not accepted and not timed out) during
unload of chan_sip, Asterisk would segfault when trying to send a signal to
a thread whose thread ID hadn't been recorded yet. This commit fixes that by
recording the thread ID before calling the blocking connect() syscall.
This was a regression introduced by 776a14386a55b5425c7e9617eff8af8b45427144.
The above wasn't enough to fix the segfault, which was now delayed to the
point where connect() timed out. Therefore, it was necessary to also remove
the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be
used to interruput the connect() syscall.
This was a regression introduced by 5d313f51b982a18f7321adcf7c7a4e822d8b2714.
ASTERISK-26586 #close
Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b
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Libedit 3.1 is not build with unicode on as a default and so the
prototype for the el_gets callback changed from expecting a char buffer
to accepting a wchar buffer. If ast_el_read_char isn't changed,
the cli reads garbage from teh terminal.
Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and
updated ast_el_read_char to use the HAVE_ define to detemrine whether
to use char or wchar.
ASTERISK-26592 #close
Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a
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The dynamic range (96-127) allows 32 RTP Payload Types. RFC 3551 section 3
allows to reassign other ranges. Consequently, when the dynamic range is
exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in asterisk.conf. This
enables the range 35-63 (or 0-63) giving room for another 29 (or 64) payload
types.
ASTERISK-26311 #close
Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964
(cherry picked from commit 9ac53877f688c06acaa7c377f15da8770e4ee88b)
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It is only safe to run ast_register_cleanup callbacks when all modules
have been unloaded. Previously these callbacks were run during graceful
shutdown, making it possible to crash during shutdown.
ASTERISK-26513 #close
Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21
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restart." into 13
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Move the function outside the conditional block that excludes
LOW_MEMORY.
ASTERISK-26273 #close
Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4
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sd_notify() is used to notify systemd of changes to the status of the
process. This allows the systemd daemon to know when the process
finished loading (and thus only start another program after Asterisk has
finished loading).
To use this, use a systemd unit with 'Type=notify' for Asterisk.
This commit also adds the function ast_sd_notify(), a wrapper around
sd_notify that does nothing if not built with systemd support.
Also adds support for libsystemd detection in the configure script.
Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811
(cherry picked from commit 07b95f7c65b7c083724f1af2b26f93cc22cad58c)
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Without this change, a 'core restart' would kill the astcanary forever
if you're not running as root. Both with and without this patch, the
scheduling priority was still SCHED_RR after restart.
Additionally, the astcanary is now spawned if you start with high
priority and Asterisk doesn't get a chance to lower it. For example
through: `chrt -r 10 sudo -u asterisk asterisk -c`
Also reap killed astcanary processes on core restart.
ASTERISK-26352 #close
Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55
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threads." into 13
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Previously only the canary checking thread itself had its priority set
to SCHED_OTHER. Now all threads are traversed and adjusted.
ASTERISK-19867 #close
Reported by: Xavier Hienne
Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39
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If sysinfo() is available, but not sysctl() or swapctl() the
printing code for swap buffer sizes is incorrectly omitted.
The above condition happens with musl c-library.
Fix #if rule to consider defined(HAVE_SYSINFO). And also
remove the redundant || defined(HAVE_SYSCTL) which was
incorrectly there to start with. Now swap information is
displayed only if an actual libc function to get it is
available.
This also fixes warnings previously seen with musl libc:
[CC] asterisk.c -> asterisk.o
asterisk.c: In function 'handle_show_sysinfo':
asterisk.c:773:6: warning: variable 'totalswap' set but not used
[-Wunused-but-set-variable]
int totalswap = 0;
^~~~~~~~~
asterisk.c:770:11: warning: variable 'freeswap' set but not used
[-Wunused-but-set-variable]
uint64_t freeswap = 0;
^~~~~~~~
Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca
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The Exchanging Device and Mailbox States could not working
if the Entity ID (EID) is not set manually and can't be obtained
from ethernet interface.
This patch replaces debug message to warning
and addes missing description about option 'entityid' to
asterisk.conf.sample.
With this patch the asterisk also:
(1) decline loading the modules which won't work without EID:
res_corosync and res_pjsip_publish_asterisk.
(2) warn if EID is empty on loading next modules:
pbx_dundi, res_xmpp
Starting with v197 systemd/udev will automatically assign "predictable"
names for all local Ethernet interfaces.
This patch also addes some new ethernet prefixes "eno" and "ens".
ASTERISK-26164 #close
Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6
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Errors during startup result in an exit. These error branches should be
calling ast_run_atexit(0) to ensure mandatory cleanup is run.
ASTERISK-26267 #close
Change-Id: If226f2326ae2df7add20040696132214cf2bb680
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This ensures startup is canceled due to allocation failures from the
following initializations.
* channel.c: ast_channels_init
* config_options.c: aco_init
ASTERISK-26265 #close
Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611
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Upcoming features will require the generation and persistence
of a UUID.
Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d
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With CLI "core show settings", simply the parameter maxfiles of the file
asterisk.conf was shown. If that parameter was not set, nothing was displayed
although the environment might have set a default number itself. Or if maxfiles
were not granted (completely), still maxfiles was shown. Now, the maximum number
of possible file descriptors in the environment is shown.
ASTERISK-26097
Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b
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Locking some objects like sorcery objects can be tricky because the underlying
ao2 object may not be the same for all callers. For instance, two threads that
call ast_sorcery_retrieve_by_id on the same aor name might actually get 2
different ao2 objects if the underlying wizard had to rehydrate the aor from a
database. Locking one ao2 object doesn't have any effect on the other even if
those objects had locks in the first place.
Named locks allow access control by keyspace and key strings. Now an "aor"
named "1000" can be locked and any other thread attempting to lock "aor" "1000"
will wait regardless of whether the underlying ao2 object is the same or not.
Mutex and rwlocks are supported.
This capability will initially be used to lock an aor when multiple threads may
be attempting to prune expired contacts from it.
Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45
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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
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Background here:
http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html
From CHANGES:
* To help insure that Asterisk is compiled and run with the same known
version of pjproject, a new option (--with-pjproject-bundled) has been
added to ./configure. When specified, the version of pjproject specified
in third-party/versions.mak will be downloaded and configured. When you
make Asterisk, the build process will also automatically build pjproject
and Asterisk will be statically linked to it. Once a particular version
of pjproject is configured and built, it won't be configured or built
again unless you run a 'make distclean'.
To facilitate testing, when 'make install' is run, the pjsua and pjsystest
utilities and the pjproject python bindings will be installed in
ASTDATADIR/third-party/pjproject.
The default behavior remains building with the shared pjproject
installation, if any.
Building:
All you have to do is include the --with-pjproject-bundled option on
the ./configure command line (and remove any existing --with-pjproject
option if specified). Everything else is automatic.
Behind the scenes:
The top-level Makefile was modified to include 'third-party' in the
list of MOD_SUBDIRS.
The third-party directory was created to contain any third party
packages that may be needed in the future. Its Makefile automatically
iterates over any subdirectories passing on targets.
The third-party/pjproject directory was created to house the pjproject
source distribution. Its Makefile contains targets to download, patch
configure, generate dependencies, compile libs, apps and python bindings,
sanitized build.mak and generate a symbols list.
When bootstrap.sh is run, it automatically includes the configure.m4
file in third-party/pjproject. This file has a macro to download and
conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR
and PJPROJECT_BUNDLED. It also tests for the capabilities like
PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to
trying to compile. Of course, bootstrap.sh is only run once and the
configure file is incldued in the patch.
When configure is run with the new options, the macro in configure.m4
triggers the download, patch, conifgure and tests. No compilation is
performed at this time. The downloaded tarball is cached in /tmp so
it doesn't get downloaded again on a distclean.
When make is run in the top-level Asterisk source directory, it will
automatically descend all the subdirectories in third_party just as it
does for addons, apps, etc. The top-level Makefile makes sure that
the 'third-party' is built before 'main' so that dependencies from the
other directories are built first.
When main does build, a new shared library (libasteriskpj) is created that
links statically to the pjproject .a files and exports all their symbols.
The asterisk binary links to that, just as it does with libasteriskssl.
When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject
python bindings are installed in ASTDATADIR/third-party/pjproject. This
will facilitate testing, including running the testsuite which will be
updated to check that directory for the pjsua module ahead of the system
python library.
Modules should continue to depend on pjproject if they use pjproject APIs
directly. They should not care about the implementation. No changes to any
res_pjsip modules were made.
Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103
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listener() opens the same code block in two places (#if and #else). This
confuses some folding editors causing it to think that an extra code block
was opened. Folding in 'geany' causes all code after listener() to be
folded as if it were part of that procedure.
ASTERISK-24813 #close
Change-Id: I4b8c766e6c91e327dd445e8c18f8a6f268acd961
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Make sure buf[res] is not accessed at res=-1 (buffer underrun).
Address Sanitizer will complain about this quite loudly.
ASTERISK-24801 #close
Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9
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This is the sixth patch in a series meant to reduce the bulk of pbx.c.
This moves hangup handler management functions to their own source.
Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104
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This is the sixth patch in a series meant to reduce the bulk of pbx.c.
This moves dialplan application management functions to their own source.
Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c
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This is the fifth patch in a series meant to reduce the bulk of pbx.c.
This moves ast_switch functions to their own source.
Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e
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This is the third patch in a series meant to reduce the bulk of pbx.c.
This moves channel and global variable routines to their own source.
Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6
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This is the second patch in a series meant to reduce the bulk of pbx.c.
This moves custom function management routines to their own source.
Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177
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We joked about splitting pbx.c into multiple files but this first step was
fairly easy. All of the pbx_builtin dialplan applications have been moved
into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins()
is called by asterisk.c just after load_pbx().
A few functions were renamed and are cross-exposed between the 2 source files.
Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a
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Several issues are addressed here:
- main() is large, and half of it is only used if we're not rasterisk;
fixed by spliting up the daemon part into a separate function.
- Call ast_term_init from rasterisk as well.
- Remove duplicate code reading/writing asterisk history file.
- Attempt to tackle background color issues and color changes that
occur. Tested by starting asterisk -c until the colors stopped
changing at odd locations.
ASTERISK-25585 #close
Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f
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A previous commit reduced the AST_BUILDOPTS compiler define to
only include options that affected ABI. This included some options
that were previously displayed by cli "core show settings". This
change corrects the CLI display while still restricting buildopts.h
to ABI effecting options only.
ASTERISK-25434 #close
Reported by: Rusty Newton
Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325
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The code which reads asterisk.conf supports processing the debug
option with ast_true, but ast_true returns -1. This causes debug
to still be off, convert to 1 so debug will be on as requested.
ASTERISK-25042
Reported by: Corey Farrell
Change-Id: I3c898b7d082d914b057e111b9357fde46bad9ed6
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This ensures that refdebug is initialized before AO2_DEBUG if
both are enabled, since AO2_DEBUG allocates a container.
This change also makes AO2_DEBUG initialization critical, a
failure will abort Asterisk startup. This is needed since
the failure would be caused by reg_containers allocation
failure, and that would result in a segmentation fault by
ao2_container_register later in startup.
ASTERISK-25048 #close
Reported by: Corey Farrell
Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244
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Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file.
As a result, this patch removes Asterisk's support for showing source file
versions.
Specifically, it does the following:
* main/asterisk:
- Refactor the file_version structure to reflect that it no longer
tracks a version field.
- Alter the "core show file version" CLI command such that it always
reports the version of Asterisk. The file version is no longer
available.
* main/manager: The Version key now always reports the Asterisk version.
* UPGRADE: Add notes for:
- Modification to the ModuleCheck AMI Action.
- Modification of the "core show file version" CLI command.
Change-Id: Ia932d3c64cd18a14a3c894109baa657ec0a85d28
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Until we have a true module management facility it's sometimes necessary for one
module to force a reload on another before its own load is complete. If
Asterisk isn't fully booted yet, these reloads are deferred. The problem is
that asterisk reports fully booted before processing the deferred reloads which
means Asterisk really isn't quite ready when it says it is.
This patch moves the report of fully booted after the processing of the deferred
reloads is complete.
Since the pjsip stack has the most number of related modules, I ran the
channels/pjsip testsuite to make sure there aren't any issues. All tests
passed.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4604/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Since 'core stop now' and 'core restart now' do not stop modules,
it is unsafe for most of the core to run cleanups. Originally all
cleanups used ast_register_atexit, and were only changed when it
was shown to be unsafe. ast_register_atexit is now used only when
absolutely required to prevent corruption and close child processes.
Exceptions that need to use ast_register_atexit:
* CDR: Flush records.
* res_musiconhold: Kill external applications.
* AstDB: Close the DB.
* canary_exit: Kill canary process.
ASTERISK-24142 #close
Reported by: David Brillert
ASTERISK-24683 #close
Reported by: Peter Katzmann
ASTERISK-24805 #close
Reported by: Badalian Vyacheslav
ASTERISK-24881 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4500/
Review: https://reviewboard.asterisk.org/r/4501/
........
Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch addresses compilation issues for OpenBSD. Specifically, it
addresses:
* It allows including <sys/vmmeter.h> in asterisk.c
* Provides a needed (size_t) cast in xmldoc.c
In 13+, it also addresses a conditional inclusion in loader.c.
Review: https://reviewboard.asterisk.org/r/4506
ASTERISK-24880 #close
Reported by: snuffy
Tested by: snuffy
patches:
misc-openbsd.diff uploaded by snuffy (License 5024)
........
Merged revisions 433245 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Using DEBUG_CHAOS several instances of a null
pointer crash, and one uninitialized variable
were uncovered and fixed. Also added details
on why Asterisk failed to initialize.
Review: https://reviewboard.asterisk.org/r/4468/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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There are three CLI commands to stop and restart Asterisk each.
1) core stop/restart now - Hangup all calls and stop or restart Asterisk.
New channels are prevented while the shutdown request is pending.
2) core stop/restart gracefully - Stop or restart Asterisk when there are
no calls remaining in the system. New channels are prevented while the
shutdown request is pending.
3) core stop/restart when convenient - Stop or restart Asterisk when there
are no calls in the system. New calls are not prevented while the
shutdown request is pending.
ARI has made stopping/restarting Asterisk more problematic. While a
shutdown request is pending it is desirable to continue to process ARI
HTTP requests for current calls. To handle the current calls while a
shutdown request is pending, a new committed to shutdown phase is needed
so ARI applications can deal with the calls until the system is fully
committed to shutdown.
* Added a new shutdown committed phase so ARI applications can deal with
calls until the final committed to shutdown phase is reached.
* Made refuse new HTTP requests when the system has reached the final
system shutdown phase. Starting anything while the system is actively
releasing resources and unloading modules is not a good thing.
* Split the bridging framework shutdown to not cleanup the global bridging
containers when shutting down in a hurry. This is similar to how other
modules prevent crashes on rapid system shutdown.
* Moved ast_begin_shutdown(), ast_cancel_shutdown(), and
ast_shutting_down(). You should not have to include channel.h just to
access these system functions.
ASTERISK-24752 #close
Reported by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/4399/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch addresses compilation errors on OS X. It's been a while, so
there's quite a few things.
* Fixed __attribute__ decls in route.h to be portable.
* Fixed htonll and ntohll to work when they are defined as macros.
* Replaced sem_t usage with our ast_sem wrapper.
* Added ast_sem_timedwait to our ast_sem wrapper.
* Fixed some GCC 4.9 warnings using sig*set() functions.
* Fixed some format strings for portability.
* Fixed compilation issues with res_timing_kqueue (although tests still fail
on OS X).
* Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue
on OS X).
ASTERISK-24539 #close
Reported by: George Joseph
ASTERISK-24544 #close
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/4327/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The QUEUESTART log entry has historically acted like a fully booted event
for the queue_log file. When the QUEUESTART entry was posted to the log
was broken by the change made by ASTERISK-15863.
* Made post the QUEUESTART queue_log entry when Asterisk fully boots.
This restores the intent of that log entry and happens after realtime has
had a chance to load.
AST-1444 #close
Reported by: Denis Martinez
Review: https://reviewboard.asterisk.org/r/4282/
........
Merged revisions 430009 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When connecting to the remote console, an id string
is first provided that consts of the hostname, pid,
and version. This is parsed by the remote instance
using a buffer that may be too short, and can allow
a buffer overrun because it is not terminated. This
patch adds termination and a larger buffer.
Review: https://reviewboard.asterisk.org/r/4182/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
1. Asterisk was limited in how many formats it could handle.
2. Formats, being a bit field, could not include any attribute information.
A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.
Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.
Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.
Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.
Generally, the new philosophy for handling formats is as follows:
* The ast_format structure is reference counted. This removed a large amount
of the memory allocations and copying that was done in prior versions.
* In order to prevent race conditions while keeping things performant, the
ast_format structure is immutable by convention and lock-free. Violate this
tenet at your peril!
* Because formats are reference counted, codecs are also reference counted.
The Asterisk core generally provides built-in codecs and caches the
ast_format structures created to represent them. Generally, to prevent
inordinate amounts of module reference bumping, codecs and formats can be
added at run-time but cannot be removed.
* All compatibility with the bit field representation of codecs/formats has
been moved to a compatibility API. The primary user of this representation
is chan_iax2, which must continue to maintain its bit-field usage of formats
for interoperability concerns.
* When a format is negotiated with attributes, or when a format cannot be
represented by one of the cached formats, a new format object is created or
cloned from an existing format. That format may have the same codec
underlying it, but is a different format than a version of the format with
different attributes or without attributes.
* While formats are reference counted objects, the reference count maintained
on the format should be manipulated with care. Formats are generally cached
and will persist for the lifetime of Asterisk and do not explicitly need
to have their lifetime modified. An exception to this is when the user of a
format does not know where the format came from *and* the user may outlive
the provider of the format. This occurs, for example, when a format is read
from a channel: the channel may have a format with attributes (hence,
non-cached) and the user of the format may last longer than the channel (if
the reference to the channel is released prior to the format's reference).
For more information on this work, see the API design notes:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.
There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).
Reviews:
https://reviewboard.asterisk.org/r/3814
https://reviewboard.asterisk.org/r/3808
https://reviewboard.asterisk.org/r/3805
https://reviewboard.asterisk.org/r/3803
https://reviewboard.asterisk.org/r/3801
https://reviewboard.asterisk.org/r/3798
https://reviewboard.asterisk.org/r/3800
https://reviewboard.asterisk.org/r/3794
https://reviewboard.asterisk.org/r/3793
https://reviewboard.asterisk.org/r/3792
https://reviewboard.asterisk.org/r/3791
https://reviewboard.asterisk.org/r/3790
https://reviewboard.asterisk.org/r/3789
https://reviewboard.asterisk.org/r/3788
https://reviewboard.asterisk.org/r/3787
https://reviewboard.asterisk.org/r/3786
https://reviewboard.asterisk.org/r/3784
https://reviewboard.asterisk.org/r/3783
https://reviewboard.asterisk.org/r/3778
https://reviewboard.asterisk.org/r/3774
https://reviewboard.asterisk.org/r/3775
https://reviewboard.asterisk.org/r/3772
https://reviewboard.asterisk.org/r/3761
https://reviewboard.asterisk.org/r/3754
https://reviewboard.asterisk.org/r/3753
https://reviewboard.asterisk.org/r/3751
https://reviewboard.asterisk.org/r/3750
https://reviewboard.asterisk.org/r/3748
https://reviewboard.asterisk.org/r/3747
https://reviewboard.asterisk.org/r/3746
https://reviewboard.asterisk.org/r/3742
https://reviewboard.asterisk.org/r/3740
https://reviewboard.asterisk.org/r/3739
https://reviewboard.asterisk.org/r/3738
https://reviewboard.asterisk.org/r/3737
https://reviewboard.asterisk.org/r/3736
https://reviewboard.asterisk.org/r/3734
https://reviewboard.asterisk.org/r/3722
https://reviewboard.asterisk.org/r/3713
https://reviewboard.asterisk.org/r/3703
https://reviewboard.asterisk.org/r/3689
https://reviewboard.asterisk.org/r/3687
https://reviewboard.asterisk.org/r/3674
https://reviewboard.asterisk.org/r/3671
https://reviewboard.asterisk.org/r/3667
https://reviewboard.asterisk.org/r/3665
https://reviewboard.asterisk.org/r/3625
https://reviewboard.asterisk.org/r/3602
https://reviewboard.asterisk.org/r/3519
https://reviewboard.asterisk.org/r/3518
https://reviewboard.asterisk.org/r/3516
https://reviewboard.asterisk.org/r/3515
https://reviewboard.asterisk.org/r/3512
https://reviewboard.asterisk.org/r/3506
https://reviewboard.asterisk.org/r/3413
https://reviewboard.asterisk.org/r/3410
https://reviewboard.asterisk.org/r/3387
https://reviewboard.asterisk.org/r/3388
https://reviewboard.asterisk.org/r/3389
https://reviewboard.asterisk.org/r/3390
https://reviewboard.asterisk.org/r/3321
https://reviewboard.asterisk.org/r/3320
https://reviewboard.asterisk.org/r/3319
https://reviewboard.asterisk.org/r/3318
https://reviewboard.asterisk.org/r/3266
https://reviewboard.asterisk.org/r/3265
https://reviewboard.asterisk.org/r/3234
https://reviewboard.asterisk.org/r/3178
ASTERISK-23114 #close
Reported by: mjordan
media_formats_translation_core.diff uploaded by kharwell (License 6464)
rb3506.diff uploaded by mjordan (License 6283)
media_format_app_file.diff uploaded by kharwell (License 6464)
misc-2.diff uploaded by file (License 5000)
chan_mild-3.diff uploaded by file (License 5000)
chan_obscure.diff uploaded by file (License 5000)
jingle.diff uploaded by file (License 5000)
funcs.diff uploaded by file (License 5000)
formats.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
bridges.diff uploaded by file (License 5000)
mf-codecs-2.diff uploaded by file (License 5000)
mf-app_fax.diff uploaded by file (License 5000)
mf-apps-3.diff uploaded by file (License 5000)
media-formats-3.diff uploaded by file (License 5000)
ASTERISK-23715
rb3713.patch uploaded by coreyfarrell (License 5909)
rb3689.patch uploaded by mjordan (License 6283)
ASTERISK-23957
rb3722.patch uploaded by mjordan (License 6283)
mf-attributes-3.diff uploaded by file (License 5000)
ASTERISK-23958
Tested by: jrose
rb3822.patch uploaded by coreyfarrell (License 5909)
rb3800.patch uploaded by jrose (License 6182)
chan_sip.diff uploaded by mjordan (License 6283)
rb3747.patch uploaded by jrose (License 6182)
ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
sip_cleanup.diff uploaded by opticron (License 6273)
chan_sip_caps.diff uploaded by mjordan (License 6283)
rb3751.patch uploaded by coreyfarrell (License 5909)
chan_sip-3.diff uploaded by file (License 5000)
ASTERISK-23960 #close
Tested by: opticron
direct_media.diff uploaded by opticron (License 6273)
pjsip-direct-media.diff uploaded by file (License 5000)
format_cap_remove.diff uploaded by opticron (License 6273)
media_format_fixes.diff uploaded by opticron (License 6273)
chan_pjsip-2.diff uploaded by file (License 5000)
ASTERISK-23966 #close
Tested by: rmudgett
rb3803.patch uploaded by rmudgetti (License 5621)
chan_dahdi.diff uploaded by file (License 5000)
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
rb3814.patch uploaded by rmudgett (License 5621)
moh_cleanup.diff uploaded by opticron (License 6273)
bridge_leak.diff uploaded by opticron (License 6273)
translate.diff uploaded by file (License 5000)
rb3795.patch uploaded by rmudgett (License 5621)
tls_fix.diff uploaded by mjordan (License 6283)
fax-mf-fix-2.diff uploaded by file (License 5000)
rtp_transfer_stuff uploaded by mjordan (License 6283)
rb3787.patch uploaded by rmudgett (License 5621)
media-formats-explicit-translate-format-3.diff uploaded by file (License 5000)
format_cache_case_fix.diff uploaded by opticron (License 6273)
rb3774.patch uploaded by rmudgett (License 5621)
rb3775.patch uploaded by rmudgett (License 5621)
rtp_engine_fix.diff uploaded by opticron (License 6273)
rtp_crash_fix.diff uploaded by opticron (License 6273)
rb3753.patch uploaded by mjordan (License 6283)
rb3750.patch uploaded by mjordan (License 6283)
rb3748.patch uploaded by rmudgett (License 5621)
media_format_fixes.diff uploaded by opticron (License 6273)
rb3740.patch uploaded by mjordan (License 6283)
rb3739.patch uploaded by mjordan (License 6283)
rb3734.patch uploaded by mjordan (License 6283)
rb3689.patch uploaded by mjordan (License 6283)
rb3674.patch uploaded by coreyfarrell (License 5909)
rb3671.patch uploaded by coreyfarrell (License 5909)
rb3667.patch uploaded by coreyfarrell (License 5909)
rb3665.patch uploaded by mjordan (License 6283)
rb3625.patch uploaded by coreyfarrell (License 5909)
rb3602.patch uploaded by coreyfarrell (License 5909)
format_compatibility-2.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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It's been 2014 for like... 6 months.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.
Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.
The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.
For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.
And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.
To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.
Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.
We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.
It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.
And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.
Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.
This patch removes:
* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge
It removes the following applications/functions:
* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO
It removes the colon delimiter from the SIPPEER function.
Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.
Review: https://reviewboard.asterisk.org/r/3698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch addresses some aesthetic issues in Asterisk. These are all just
minor tweaks to improve the look of the CLI when used in a variety of
settings. Specifically:
* A number of chatty verbose messages were removed or demoted to DEBUG
messages. Verbose messages with a verbosity level of 5 or higher were -
if kept as verbose messages - demoted to level 4. Several messages
that were emitted at verbose level 3 were demoted to 4, as announcement
of dialplan applications being executed occur at level 3 (and so the
effects of those applications should generally be less).
* Some verbose messages that only appear when their respective 'debug'
options are enabled were bumped up to always be displayed.
* Prefix/timestamping of verbose messages were moved to the verboser
handlers. This was done to prevent duplication of prefixes when the
timestamp option (-T) is used with the CLI.
* Verbose magic is removed from messages before being emitted to
non-verboser handlers. This prevents the magic in multi-line verbose
messages (such as SIP debug traces or the output of DumpChan) from
being written to files.
* _Slightly_ better support for the "light background" option (-W) was
added. This includes using ast_term_quit in the output of XML
documentation help, as well as changing the "Asterisk Ready" prompt to
bright green on the default background (which stands a better chance of
being displayed properly than bright white).
Review: https://reviewboard.asterisk.org/r/3547/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
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Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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