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
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
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 patch causes a logger message to be the same as it is in 15+. This
will allow a follow-up patch to be cherry-picked to all 3 branches.
Change-Id: Ic0665a3d49987e4eb6df28dcd9e90b1c3ca191e0
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
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
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.
Patches against 13 and 15 have a compatibility layer needed to
maintain ABI that is not needed in master.
Change-Id: I56f4e20ba1154bd52281f995c27a429a854f6a79
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
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
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
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
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
* 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
The configure option to disable XML documentation does not currently
work. This patch makes it effective, but also causes an ABI change by
removing the ast_xmldoc_* symbols. Disabling xmldoc also prevents docs
from being automatically generated, but they can still be manually
generated with 'make doc/core-en_US.xml'.
ASTERISK-26639
Change-Id: Ifac562340c09f80c83e0203de098fcac93bf8c44
Currently ast_http_send barricades a portion of the content that
needs to be sent in order to establish a connection for things
like the ARI client. The conditional and contents have been changed
to ensure that everything that needs to be sent, will be sent.
ASTERISK-27372
Change-Id: I8816d2d8f80f4fefc6dcae4b5fdfc97f1e46496d
* Rename the Party A CDR container from active_cdrs_by_channel to
active_cdrs_master.
* Renamed the support functions associated with active_cdrs_master
appropriately.
ASTERISK-27335
Change-Id: I6104bb3edc3a0b7243ce502e45e8832b0cff14f7
The CDR performance gets worse the further it gets behind in processing
stasis messages. One of the reasons is because of a n*m loop used when
processing Party B information.
* Added a new CDR container that is keyed to Party B so we don't need such
a large loop when processing Party B information.
NOTE: To reduce the size of the patch I deferred to another patch the
renaming of the Party A active_cdrs_by_channel container to
active_cdrs_master and renaming the container's hash and cmp functions
appropriately.
ASTERISK-27335
Change-Id: I0bf66e8868f8adaa4b5dcf9e682e34951c350249
It's possible for bfdobj to be created but syms not created. If syms
was not allocated in the current loop iteration but was allocated in the
previous iteration it would crash.
ASTERISK-27340
Change-Id: I5b110c609f6dfe91339f782a99a431bca5837363
This avoids a crash on stopping a chan_sip which failed to start its TLS server.
ASTERISK-27339 #close
Change-Id: I327fc70db68eaaca5b50a15c7fd687fde79263d5
Corey Farrell
Richard Mudgett
Niklas Larsson
Jenkins2
Sean Bright
Joshua Colp
George Joseph
Alexander Traud
Kevin Harwell
Ben Ford
Tzafrir Cohen