Corey Farrell [Wed, 31 Jan 2018 02:31:51 +0000 (21:31 -0500)]
json: Add conditionals to avoid locking if Jansson is thread safe.
Jansson is thread safe for all read-only functions and reference
counting starting v2.11. This allows simplification of our code and
removal of locking around reference counting and dumping.
Richard Mudgett [Wed, 31 Jan 2018 23:48:46 +0000 (17:48 -0600)]
app_confbridge: ConfbridgeList event has standard channel shapshot headers.
* Made the AMI ConfbridgeList action's ConfbridgeList events output all
the standard channel snapshot headers instead of a few hand-coded channel
snapshot headers. The benefit is that the CallerIDName gets disruptive
characters like CR, LF, Tab, and a few others escaped. However, an empty
CallerIDName is now output as "<unknown>" instead of "<no name>".
Oron Peled [Tue, 19 Dec 2017 08:52:54 +0000 (10:52 +0200)]
chan_console: don't read and write at the same time
It seems that the ALSA backend of PortAudio doesn't know how to both
read and write at the same time by adding a per-device mutex.
FIXME: currently only a draft version. Need to either auto-detect
we work with the ALSA backend or add an extra configuration option
to use this mutex.
Richard Mudgett [Fri, 2 Feb 2018 23:35:20 +0000 (17:35 -0600)]
endpoint identifiers: Some code cleanup.
res_pjsip_endpoint_identifier_user.c:
* Fix copy/paste error in find_endpoint(). We were using a constant
"anonymous" string instead of the passed in endpoint_name when checking
the transport domain for an endpoint match.
* Eliminate RAII_VAR in find_endpoint().
* Remove always true check in find_transport_state_in_use().
* Remove useless CMD_STOP in find_transport_state_in_use().
res_pjsip_endpoint_identifier_anonymous.c:
* Eliminate RAII_VAR in anonymous_identify().
* Remove always true check in find_transport_state_in_use().
* Remove useless CMD_STOP in find_transport_state_in_use().
Richard Mudgett [Wed, 31 Jan 2018 01:22:51 +0000 (19:22 -0600)]
bridge_softmix.c: Report not talking immediately when muted.
Currently in app_confbridge if someone mutes a channel while that channel
is talking, the talk detection code is suspended while the channel is
muted. As far an an external observer is concerned, the muted channel's
talk status is still "talking" even though the channel is not contributing
audio to the conference bridge. When the channel is later unmuted, it
takes the usual 'dsp_silence_threshold' option time to clear the talking
status even though the channel may have stopped talking while the channel
was muted.
* In bridge_softmix.c, clear the talking status and report talking stopped
if the channel was talking when the channel is muted. When the channel is
unmuted and the channel is still talking then report the channel as
talking since it is contributing audio to the bridge again.
Richard Mudgett [Tue, 30 Jan 2018 21:00:32 +0000 (15:00 -0600)]
app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs.
The dsp_talking_threshold does not represent time in milliseconds. It
represents the average magnitude per sample in the audio packets. This is
what the DSP uses to determine if a packet is silence or talking/noise.
George Joseph [Sun, 28 Jan 2018 16:10:00 +0000 (09:10 -0700)]
res_pjsip_pubsub: Prune subs with reliable transports at startup
In an earlier release, inbound registrations on a reliable transport
were pruned on Asterisk restart since the TCP connection would have
been torn down and become unusable when Asterisk stopped. This same
process is now also applied to inbound subscriptions.
Also fixed issues in res_pjsip_registrar where it wasn't handling the
monitoring correctly when multiple registrations came in over the same
transport.
To accomplish this, the pjsip_transport_event feature needed to
be refactored to allow multiple monitors (multiple subcriptions or
registrations from the same endpoint) to exist on the same transport.
Since this changed the API, any external modules that may have used the
transport monitor feature (highly unlikey) will need to be changed.
Corey Farrell [Mon, 29 Jan 2018 13:51:55 +0000 (08:51 -0500)]
Sample modules.conf: comment out example load statement.
The sample modules.conf explicitly loaded res_musiconhold.so. This is
redundent as autoload=yes is already set. It causes warnings if
res_musiconhold.so was not installed and results in an unexpected load
if the admin disables autoload without remembering to remove the
res_musiconhold load statement.
Also remove reference to unknown module pbx_gtkconsole.
Alexander Traud [Sun, 28 Jan 2018 11:20:26 +0000 (12:20 +0100)]
BuildSystem: Remove unused variables.
Because of a copy-and-paste from the script build_tools/download_externals,
the script build_tools/list_valid_installed_externals got its local variables.
However in the latter, three variables were not used actually.
Corey Farrell [Thu, 25 Jan 2018 18:06:12 +0000 (13:06 -0500)]
loader: Use ast_cli_completion_add for 'module load' completion.
This addresses all performance issues with 'module load' completion. In
addition to using ast_cli_completion_add we stop using libedit's
filename_completion_function, instead using ast_file_read_dir. This
ensures all results are produced from a single call to opendir.
Alexander Traud [Sat, 27 Jan 2018 09:25:11 +0000 (10:25 +0100)]
headers: Consistent use of typeof and/or __typeof__.
Because of a copy-and-paste error, the Asterisk project was using __typeof
instead of typeof. It works because typeof, __typeof, and __typeof__ are
supported by GCC, but here the escaped variant was not intended. Therefore,
for consistence, we change this to typeof.
Alexander Traud [Mon, 15 Jan 2018 17:08:24 +0000 (18:08 +0100)]
BuildSystem: Raise autoconf version requirement to 2.60a.
AC_COMPUTE_INT requires at least autoconf 2.60a.
This affects only those who contribute to Asterisk, only those who had to use
the script ./bootstrap.sh. Furthermore, this change just makes sure nobody is
using a too old autoconf.
Corey Farrell [Mon, 1 Jan 2018 21:59:57 +0000 (16:59 -0500)]
core: Tweak startup order.
Move initialization of units which do not require configuration to occur
before preload modules. This leaves only units which load config between
module preload and regular load stages.
Corey Farrell [Thu, 25 Jan 2018 04:44:09 +0000 (23:44 -0500)]
loader: Correct overly strict startup checks.
The code which handled loading modules had too many situations which
would result in halting Asterisk startup. Treat most errors as declines
instead of failures. The exception is when the module load function
returns AST_MODULE_LOAD_FAILURE or an invalid code.
Clear the missingdeps vector when appropriate to ensure the next loop
starts clean.
Corey Farrell [Thu, 25 Jan 2018 00:49:02 +0000 (19:49 -0500)]
Build System: Add support for __atomic built-in operators.
Add a check to configure.ac for __atomic_fetch_add support. If found
use the __atomic built-in operators for ast_atomic_dec_and_test and
ast_atomic_fetchadd_int.
Richard Mudgett [Fri, 19 Jan 2018 02:19:00 +0000 (20:19 -0600)]
pbx_variables.c: Misc fixes in variable substitution.
* Copy more than one character at a time when there is nothing to
substitute.
* Fix off by one error if a '}' or ']' is missing.
* Eliminated the requirement that the "used" parameter had to point to a
variable. The current callers were always declaring a variable to meet
the requirement and discarding the value put into that variable. Now it
can be NULL.
* In ast_str_substitute_variables_full() fixed using the bogus channel to
evaluate a function. We were not using the bogus channel we just created
to help evaluate a subexpression.
Corey Farrell [Thu, 18 Jan 2018 15:01:26 +0000 (10:01 -0500)]
Add missing OPTIONAL_API and ARI dependences.
I've audited all modules that include any header which includes
asterisk/optional_api.h. All modules which use OPTIONAL_API now declare
those dependencies in AST_MODULE_INFO using requires or optional_modules
as appropriate.
In addition ARI dependency declarations have been reworked. Instead of
declaring additional required modules in res/ari/resource_*.c we now add
them to an optional array "requiresModules" in api-docs for each module.
This allows the AST_MODULE_INFO dependencies to include those missing
modules.
Alexander Traud [Fri, 5 Jan 2018 20:46:51 +0000 (21:46 +0100)]
install_prereq: For PJProject, point users to configure script.
The installation script and the new configure option --with-pjproject-bundled
aimed to accomplish the same. However, the installation script was out of
date. Users should go for the maintained configure option, or the Wiki.
projects / thirdparty / asterisk.git / log
