Corey Farrell [Thu, 7 Dec 2017 16:35:39 +0000 (11:35 -0500)]
CLI: Fix remote console completion.
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.
Corey Farrell [Wed, 6 Dec 2017 20:49:32 +0000 (15:49 -0500)]
sounds_index: Avoid repeatedly reindexing.
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.
Corey Farrell [Wed, 6 Dec 2017 18:42:55 +0000 (13:42 -0500)]
media_index: Improve startup.
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.
Richard Mudgett [Wed, 6 Dec 2017 13:36:02 +0000 (07:36 -0600)]
bridge_basic.c: Update transfer diagnostic messages addendum.
* Added start DTMF transfer verbose messages.
* Made associated transfer messages use a similar message format.
* Adjusted message verbose level as requested by initial reporter.
The SuccessfulAuth using_password field was declared as a pointer to a
uint32_t when the field was later read as a uint32_t value. This resulted
in unnecessary casts and a non-portable field value reinterpret in
main/security_events.c:add_json_object(). i.e., It would work on a 32 bit
architecture but not on a 64 bit big endian architecture.
Sungtae Kim [Mon, 4 Dec 2017 09:40:18 +0000 (10:40 +0100)]
Add new object for VoicemailUserEntry
Currently, when the app_voicemail sending VoicemailUserEntry AMI event, there's
no OldMessageCount info for default.
To check the OldMessageCount info, it required IMAP_STORAGE define, but this is
not correct.
Added OldMessageCount item as a default.
More complicated direct media reinvite negotiations can result in longer
delays before direct media flows. The strictrtp learning timeout time
was too short. One log showed that the first RTP packet came in just
after three seconds.
* Increase the strictrtp learning timeout time from 1.5 to 5 seconds.
Alexander Traud [Mon, 4 Dec 2017 14:33:16 +0000 (15:33 +0100)]
res_rtp_asterisk: Correct default in sample configuration file.
With Asterisk 12 (commit 866d968), the default of "icesupport" changed to
- "yes" in the module "res_rtp_asterisk" and
- "no" in the module "chan_sip".
The latter was reflected in the sample configuration file for "sip.conf". The
former did not make it into "rtp.conf.sample".
Joshua Colp [Mon, 4 Dec 2017 00:49:14 +0000 (00:49 +0000)]
pjproject: Clean up disabling of WebRTC support.
The definition in config_site.h and the argument to the
configure script are not necessary to disable WebRTC
support. The correct argument, --disable-libwebrtc, is
already passed.
Corey Farrell [Sat, 2 Dec 2017 21:55:31 +0000 (16:55 -0500)]
autoconf: Remove use of m4_ifblank.
The m4_ifblank macro is not available on CentOS 6, reverse conditionals
to allow use of m4_ifval instead. ./bootstrap.sh was run but this patch
does not result in any difference to the generated configure script.
George Joseph [Thu, 30 Nov 2017 20:38:50 +0000 (13:38 -0700)]
AST-2017-013: chan_skinny: Call pthread_detach when sess threads end
chan_skinny creates a new thread for each new session. In trying
to be a good cleanup citizen, the threads are joinable and the
unload_module function does a pthread_cancel() and a pthread_join()
on any sessions that are active at that time. This has an
unintended side effect though. Since you can call pthread_join on a
thread that's already terminated, pthreads keeps the thread's
storage around until you explicitly call pthread_join (or
pthread_detach()). Since only the module_unload function was
calling pthread_join, and even then only on the ones active at the
tme, the storage for every thread/session ever created sticks
around until asterisk exits.
* A thread can detach itself so the session_destroy() function
now calls pthread_detach() just before it frees the session
memory allocation. The module_unload function still takes care
of the ones that are still active should the module be unloaded.
Sean Bright [Fri, 1 Dec 2017 16:01:01 +0000 (11:01 -0500)]
config: Speed up config template lookup
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.
Sean Bright [Fri, 1 Dec 2017 14:29:43 +0000 (09:29 -0500)]
config: Speed up ACO & sorcery initialization
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.
Alexander Traud [Sun, 26 Nov 2017 17:47:17 +0000 (18:47 +0100)]
translate: Transcode siren14, speex32, silk24, and silk12 via slin16.
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).
Corey Farrell [Thu, 23 Nov 2017 00:37:44 +0000 (19:37 -0500)]
Add defaultbranch to .gitreview.
This will cause `git review` to post changes to the branch it is based
on instead of always using master. The defaultbranch setting should be
updated when new major branches are created.
Kevin Harwell [Wed, 22 Nov 2017 16:42:48 +0000 (10:42 -0600)]
pjsip: 183 without To tag does not negotiate media
If a 183 with sdp response is receive without a To tag the sdp is not
negotiated. According to RFC 3261 section 12.1.2 while a To tag is required,
the client needs to still be able to handle the missing tag case for
backwards compatibility.
This patch, accepted by and applied to pjproject, makes it so if an incoming
180/183 with SDP comes in without a To tag it gets appropriately handled.
Alexander Traud [Tue, 21 Nov 2017 12:39:34 +0000 (13:39 +0100)]
res_rtp_asterisk: ICE server-reflexive candidates (srflx) with Dual-Stack.
Previously, Asterisk sent srflx only when configured exclusively for IPv4. Now,
srflx is gathered and sent via SDP, even when Asterisk is enabled for
Dual Stack (IPv4+IPv6) and an IPv4 interface is available/used.
Corey Farrell [Mon, 20 Nov 2017 19:05:46 +0000 (14:05 -0500)]
res_parking: Make load_pri explicit.
res_parking has an implicit load_pri of 0 meaining it's one of the very
first modules loaded after modules with global symbols. Set it
explicitly in the AST_MODULE_INFO block.
Corey Farrell [Tue, 21 Nov 2017 15:16:24 +0000 (10:16 -0500)]
README: Convert to README.md.
Convert the README file to markdown format, remove the old README. This
causes websites like github to display the README in a much nicer
format with live links. The raw file is still very readable from
plain text editors and terminals.
Corey Farrell [Tue, 7 Nov 2017 21:34:40 +0000 (16:34 -0500)]
CLI: Refactor cli_complete.
* 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.
Corey Farrell [Tue, 7 Nov 2017 20:47:51 +0000 (15:47 -0500)]
CLI: Refactor ast_cli_display_match_list.
* 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.
Corey Farrell [Thu, 9 Nov 2017 05:42:10 +0000 (00:42 -0500)]
CLI: Create ast_cli_completion_add function.
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.
Corey Farrell [Thu, 9 Nov 2017 06:39:13 +0000 (01:39 -0500)]
CLI: Remove calls to ast_cli_generator.
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.
res_mwi_external_ami specified AST_MODFLAG_LOAD_ORDER but didn't set
load_pri, resulting in an actual load priority of 0. This module only
provides AMI actions so it has no reason to load early.
projects / thirdparty / asterisk.git / log
