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
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
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
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
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
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.
Change-Id: I3db009217c5ae399fb84bee95076f4dbb7fa52d2
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.
ASTERISK-27442 #close
Change-Id: Ic9d6b01e05e8f4874eebbd7adfe05d932025d203
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.
ASTERISK-27437
Change-Id: Ie07d8e2bfa7b6fe06fcdc73d390a7a9a4d8c0bc1
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.
Change-Id: I297b6fb3ff6993ec004e667b22a74f5925906259
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.
Change-Id: I7d13131764a9a9026e5f8a6ddb245a01bbd788e7
* 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
