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Duplicate checking was done incorrectly when parsing completion options
from a remote console causing all options to be ignored as duplicates.
Once fixed I had to separate processing of the best match to ensure it
was not identified as a duplicate when it is the only match.
ASTERISK-27465
Change-Id: Ibbdb29f88211742071836c9b3f4d2aa1221cd0f9
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Change-Id: I1e5eef4029cba56e33d786c5a5ade8091e531a1e
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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
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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
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* 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
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* 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
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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
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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
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slin16." into 13
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table." into 13
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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
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ASTERISK-24662
Change-Id: I3822956984292c99c48bca8e97807e498ccc0e88
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Change-Id: Ib81318f4ee52a5e73b003316e13fe9be1dd897a1
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* 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
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This rewrites ast_el_strtoarr to use vector's internally, but still
return the original NULL terminated array of strings.
Change-Id: Ibfe776cbe14f750effa9ca360930acaccc02e957
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* 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
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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
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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
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Make the comments follow doxygen format, move comments to the line
before each field they describe.
Change-Id: Ic445468398b5e88f13910f7c2f70bd15aad33a27
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* changes:
acl: Fix allocation related issues.
acl: Update logger message to match 15+.
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Change-Id: I81aacfee7cd26e4fc5eef07bca582700c2975bd7
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out." into 13
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Change-Id: I5ce40035e0a940e4e56f6322c1dcd47fbd509b98
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Change-Id: Ideb594f7aae134974fb78d5477ba0853b97b8625
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Change-Id: I07907f833b81aeb0128bc9442a2abb52679c7511
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Change-Id: I9d56175369363d1dc735504cf78a3a5577069f49
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* 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
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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
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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
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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
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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
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