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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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Setting channel variables with the AMI Originate action caused a deadlock
when you set CDR(amaflags) or CDR(accountcode). This path has the channel
locked when the CDR function is called. The CDR function then
synchronously passes the job to a stasis thread. The stasis handling
function then attempts to lock the channel. Deadlock results.
* Avoid deadlock by making the CDR function handle setting amaflags and
accountcode directly on the channel rather than passing it off to the CDR
processing code under a stasis thread to do it.
* Made the CHANNEL function and the CDR function process amaflags the same
way.
* Fixed referencing the wrong message type in cdr_prop_write().
ASTERISK-27460
Change-Id: I5eacb47586bc0b8f8ff76a19bd92d1dc38b75e8f
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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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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.
Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935
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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.
ASTERISK-27456
Change-Id: I5c71521c2d1daf8b7b161e31c34d28cca6aea4c7
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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.
ASTERISK-27453
Change-Id: Ic5e711164cbb91b4d1c1e40c83697755640f138c
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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".
ASTERISK-20643
Change-Id: I2a2e0a900455d0767a99ea576e30adc6d7608a36
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into 15
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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.
ASTERISK-26980
Change-Id: I27da2c894f87914956a72710222e17462d8a44bc
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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.
Change-Id: I280785deb872ed8d3339d99cce63a2b54d5f1438
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Follow-up to conversion of README.md.
Change-Id: I17ee7cf25bc027ece844efa2c1dfe613aff1e35b
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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.
ASTERISK-27452
Reported by: Juan Sacco
Change-Id: I9af7268eba14bf76960566f891320f97b974e6dd
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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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This change makes the presence of the GMIME_MAJOR_VERSION
definition optional, as not all versions of gmime actually
define it.
ASTERISK-27454
Change-Id: I01d99590045971ed6787899147170a5954077238
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slin16." into 15
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table." into 15
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Change-Id: I530c0a72f965437acef6a9a4fbfe5c487f078b65
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The `pwd` parameter to AC_CONFIG_AUX_DIR is unnecessary, the default
value is $srcdir.
Additionally remove the AC_REVISION call. It only added a comment and
is pointless without SVN tag replacements.
Change-Id: I99299a3217f095bddcb2edefb3b9af0ab147bc29
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clients" into 15
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into 15
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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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The patch for ASTERISK_24560 inverted a test checking if the bridge name
is being updated to a different name.
* Fix the test to return "Changing bridge name is not implemented" when
someone attempts to change the bridge name.
ASTERISK-27445
Change-Id: I4b70bf08b0e02e016108b077ff75b345dec12fc9
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ASTERISK-24662
Change-Id: I3822956984292c99c48bca8e97807e498ccc0e88
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Change-Id: Id2899331fe05d1909a862ea879742879d086bc64
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