Corey Farrell [Fri, 16 Feb 2018 23:58:35 +0000 (18:58 -0500)]
BuildSystem: Use single bootstrap.sh for Asterisk and menuselect.
This causes the root bootstrap.sh script to generate configure scripts
for both Asterisk and menuselect. This ensures that both configure
scripts are generated with the same version of autotools and avoids
situations where shared autoconf macros get modified without
regenerating the menuselect script.
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.
Alexander Traud [Mon, 12 Feb 2018 12:16:00 +0000 (13:16 +0100)]
pjproject_bundled: Disable G.729 from Belledonne Communications.
When <http://github.com/BelledonneCommunications/bcg729> is installed, PJProject
tries to link that. Support for this bcg729 was added with PJProject 2.7. The
issue happens, because Teluu enabled that new feature on default.
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 [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>".
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().
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 [Mon, 29 Jan 2018 16:20:39 +0000 (17:20 +0100)]
BuildSystem: Enable autotools in FreeBSD.
In the current versions of FreeBSD, the apps of GNU autotools do not need to
be called with a version anymore. The latest version can be invoked directly.
Additionally, the script ./bootstrap.sh asked for autoconf 2.62 and
automake 1.9, versions which are not available as port anymore.
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.
projects / thirdparty / asterisk.git / log
