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.
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.
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.
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.
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.
Alexander Traud [Fri, 19 Jan 2018 10:46:21 +0000 (11:46 +0100)]
install_prereq: Support package manager DNF and yum option strict=1.
This re-enables the script ./contrib/scripts/install_prereq on Fedora 22 and
newer, and on RHEL/CentOS when the option strict=1 was set for yum install.
ASTERISK-27598
Reported by: Hunter Stevens, Said Masoud
krells [Tue, 9 Jan 2018 17:29:48 +0000 (18:29 +0100)]
pbx: Reduce verbosity while loading extensions
Each time the dial plan is reloaded, a lot of logs like these are generated:
"Added extension 'XXXXX' priority 1 to YYYYYYYYYYY"
This patch changes the log level for those logs.
chan_unistim: Fix hold function ability to lock/crash asterisk
This patch fix chan_unistim hold functions to correctly support
hold function in different states possible in case of multiple lines
established on the phone
Corey Farrell [Wed, 17 Jan 2018 06:28:57 +0000 (01:28 -0500)]
loader: Miscellaneous fixes.
* Remove comment about lazy load.
* Improve message about module already being loaded and running.
* Handle allocation error in add_to_load_order.
* Dead code elimination from modules_shutdown.
Alexander Traud [Wed, 17 Jan 2018 08:51:29 +0000 (09:51 +0100)]
BuildSystem: Detect external library Lua in version 5.3.
On some platforms, you decide to go for one specific version of Lua, for
example in OpenBSD. On other platforms, you are able to install several versions
side-by-side, for example in Ubuntu and Fedora. Asterisk already works with
Lua 5.3. Asterisk failed to detect Lua 5.3 on those platforms which allow
several versions.
Richard Mudgett [Tue, 16 Jan 2018 14:32:20 +0000 (08:32 -0600)]
taskprocessor.c: Increase the number of tps_singletons container buckets.
Since v12 the number of taskprocessors in the system has increased a lot.
Small systems can easily have over a hundred and larger systems can have
thousands.
Most uses of the tps_singletons container deal with creating and
destroying the taskprocessors. However, the pjsip distributor looks up
taskprocessors/serializers by name frequently. It needs to find the
serializer for incoming SIP responses to distribute them to the
appropriate serializer.
George Joseph [Tue, 16 Jan 2018 14:20:28 +0000 (07:20 -0700)]
pjproject_bundled: Prevent crash on bad outgoing header
We still need to figure out how a bad header is getting into the
outgoing message but this patch to pjproject prevents attempting
to print that header and causing a crash.
For several users, this crash happens when sending 183 progress
messages.
Asterisk's makefile for menuselect has a very simple source file parsing
script that looks for AST_MODULE_INFO lines to extract the quoted string
as a module description. If it does not find a quoted string it uses the
whole line as the description.
Corey Farrell [Sat, 13 Jan 2018 00:37:43 +0000 (19:37 -0500)]
res_stasis_recording: Allow symbolic links in configured recordings dir.
If any component of ast_config_AST_RECORDING_DIR is a symbolic link we
would incorrectly assume the ARI user was trying to escape the recording
path. Create additional check to check the recording directory's
realpath, only deny access if both do not match.
This is needed by the testsuite when run by 'run-local'.
projects / thirdparty / asterisk.git / log
