George Joseph [Tue, 12 Mar 2019 18:25:33 +0000 (12:25 -0600)]
Makefile.moddir_rules: Pass PJPROJECT_BUNDLED to download_externals
The download_externals script wasn't getting the PJPROJECT_BUNDLED
environment variable passed down to it so it wasn't downloading
the appropriate variant of res_digium_phone. This could cause
crashes in the DPMA.
sungtae kim [Sun, 3 Mar 2019 15:20:24 +0000 (16:20 +0100)]
res/res_rtp_asterisk.c: Fixing possible divide by zero
Currently, when the Asterisk calculates rtp statistics, it uses
sample_count as a unsigned integer parameter. This would be fine
for most of cases, but in case of large enough number of sample_count,
this might be causing the divide by zero error.
Torrey Searle [Mon, 4 Mar 2019 07:50:18 +0000 (08:50 +0100)]
chan_pjsip: add a flag to ignore 183 responses if no SDP present
chan_sip will always ignore 183 responses that do not contain SDP
however, chan_pjsip will currently always translate it into a
183 with SDP. This new flag allows chan_pjsip to have the same
behavior as chan_sip.
Ben Ford [Thu, 28 Feb 2019 18:03:43 +0000 (12:03 -0600)]
res_stasis: Add ability to switch applications.
Added the ability to move between Stasis applications within Stasis.
This can be done by calling 'move' in an application, providing (at
minimum) the channel's id and the application to switch to. If the
application is not registered or active, nothing will happen and the
channel will remain in the current application, and an event will be
triggered to let the application know that the move failed. The event
name is "ApplicationMoveFailed", and provides the "destination" that the
channel was attempting to move to, as well as the usual channel
information. Optionally, a list of arguments can be passed to the
function call for the receiving application. A full example of a 'move'
call would look like this:
client.channels.move(channelId, app, appArgs)
The control object used to control the channel in Stasis can now switch
which application it belongs to, rather than belonging to one Stasis
application for its lifetime. This allows us to use the same control
object instead of having to tear down the current one and create
another.
Sean Bright [Mon, 4 Mar 2019 22:05:30 +0000 (17:05 -0500)]
app_queue: Handle empty 'interface' in queue member config
While the 'interface' column is a NOT NULL, the empty string is still
allowed. res_config_odbc treats the empty string as a NULL and we crash
when trying to dereference.
Also cleaned up an adjacent error message for consistency.
Joshua Colp [Thu, 28 Feb 2019 12:24:59 +0000 (12:24 +0000)]
basic-pbx: Update configuration to work with current modules.
The res_pjsip_websocket module requires the res_http_websocket
module so ensure it is loaded. As well the res_pjsip_notify
module needs the pjsip_notify.conf configuration file so
ensure it is installed.
sungtae kim [Fri, 8 Feb 2019 21:32:18 +0000 (22:32 +0100)]
bridging: Add creation timestamps
This small feature will help to checking the bridge's status to
figure out which bridge is in old/zombie or not. Also added
detail items for the 'bridge show *' cli to provide more detail
info. And added creation item to the ARI as well.
Sean Bright [Thu, 28 Feb 2019 01:09:03 +0000 (20:09 -0500)]
res_config_odbc: Avoid deadlock when max_connections = 1
Rather than calling ast_odbc_find_table() in the prepare callback, call
it beforehand and pass it in to the callback to avoid the need for a
second connection.
Kevin Harwell [Wed, 13 Feb 2019 21:24:41 +0000 (15:24 -0600)]
res_pjsip_registrar: blocked threads on reliable transport shutdown take 3
When a contact was removed by the registrar it did not always check to see if
the circumstances involved a monitored reliable transport. For instance, if the
'remove_existing' option was set to 'true' then when existing contacts were
removed due to 'max_contacts' being reached, those existing contacts being
removed did not unregister the transport monitor.
Also, it was possible to add more than one monitor on a reliable transport for
a given aor and contact.
This patch makes it so all contact removals done by the registrar also remove
any associated transport monitors if necessary. It also makes it so duplicate
monitors cannot be added for a given transport.
George Joseph [Wed, 27 Feb 2019 16:37:14 +0000 (09:37 -0700)]
CI: Update jenkinsfiles with new Gerrit URLs
The recent upgrade of Gerrit to 2.16 elimiated referencing a
repository in a way the jenkinsfiles were relying on so
the URL references were changed to a more consistent and supported
format.
Torrey Searle [Mon, 25 Feb 2019 15:41:44 +0000 (16:41 +0100)]
res/res_rtp_asterisk: smoother can cause wrong timestamps if dtmf happen
Delivery timeval in the smoother object will fall behind while a DTMF is
being generated. This can eventually lead to invalid rtp timestamps.
To prevent this from happening the smoother needs to be reset after every
DTMF to keep the timing up to date.
Kevin Harwell [Mon, 25 Feb 2019 21:32:27 +0000 (15:32 -0600)]
rest-api-templates/asterisk_processor - replace http line breaks with line feed
Including line breaks (<br>, <br/>, <br />) in certain parts of the rest-api
json definition (e.g. summary, notes) displays them correctly in swagger.
However, when the field gets converted to the wiki format those breaks get
escaped and show up in the text as the actual string literal "<br>" etc...
This patch makes it so when converting to the wiki format it replaces all line
break values (<br>, etc...) with line feeds ('\n').
Sungtae Kim [Wed, 9 Jan 2019 10:27:03 +0000 (10:27 +0000)]
http.c: Support separated HTTP request
Currently, the Asterisk does not support seperated HTTP request.
This patch make the Asterisk enables to wait lest part of HTTP request.
Also increases acceptable HTTP body length to 40k to support more
larger request.
George Joseph [Wed, 20 Feb 2019 18:48:25 +0000 (11:48 -0700)]
Core: Increase AST_PBX_MAX_STACK to 512 if not LOW_MEMORY
The current settings AST_PBX_MAX_STACK is 128 entries which is
too low for some FreePBX installations with complex parking
arrangements. Increased to 512 if LOW_MEMORY is not defined.
Joshua C. Colp [Wed, 20 Feb 2019 18:22:31 +0000 (14:22 -0400)]
stasis: Store subscriber uniqueids with topic statistics.
This change provides an easier mechanism to determine which
subscribers are subscribed to a topic. Using this you can
inspect the specific subscribers for further details.
George Joseph [Fri, 15 Feb 2019 18:53:50 +0000 (11:53 -0700)]
taskprocessor: Enable subsystems and overload by subsystem
To prevent one subsystem's taskprocessors from causing others
to stall, new capabilities have been added to taskprocessors.
* Any taskprocessor name that has a '/' will have the part
before the '/' saved as its "subsystem".
Examples:
"sorcery/acl-0000006a" and "sorcery/aor-00000019"
will be grouped to subsystem "sorcery".
"pjsip/distributor-00000025" and "pjsip/distributor-00000026"
will bn grouped to subsystem "pjsip".
Taskprocessors with no '/' have an empty subsystem.
* When a taskprocessor enters high-water alert status and it
has a non-empty subsystem, the subsystem alert count will
be incremented.
* When a taskprocessor leaves high-water alert status and it
has a non-empty subsystem, the subsystem alert count will be
decremented.
* A new api ast_taskprocessor_get_subsystem_alert() has been
added that returns the number of taskprocessors in alert for
the subsystem.
* A new CLI command "core show taskprocessor alerted subsystems"
has been added.
* A new unit test was addded.
REMINDER: The taskprocessor code itself doesn't take any action
based on high-water alerts or overloading. It's up to taskprocessor
users to check and take action themselves. Currently only the pjsip
distributor does this.
* A new pjsip/global option "taskprocessor_overload_trigger"
has been added that allows the user to select the trigger
mechanism the distributor uses to pause accepting new requests.
"none": Don't pause on any overload condition.
"global": Pause on ANY taskprocessor overload (the default and
current behavior)
"pjsip_only": Pause only on pjsip taskprocessor overloads.
* The core pjsip pool was renamed from "SIP" to "pjsip" so it can
be properly grouped into the "pjsip" subsystem.
* stasis taskprocessor names were changed to "stasis" as the
subsystem.
* Sorcery core taskprocessor names were changed to "sorcery" to
match the object taskprocessors.
Kevin Harwell [Fri, 8 Feb 2019 19:07:13 +0000 (13:07 -0600)]
ARI event type filtering
Event type filtering is now enabled, and configurable per application. An app is
now able to specify which events are sent to the application by configuring an
allowed and/or disallowed list(s). This can be done by issuing the following:
PUT /applications/{applicationName}/eventFilter
And then enumerating the allowed/disallowed event types as a body parameter.
sungtae kim [Thu, 14 Feb 2019 23:09:30 +0000 (00:09 +0100)]
chan_pjsip: Changed to continued after invalid media for pjsip show channelstats
Currently, the pjsip show channelstats cli does not show channel's
stats after hits the invalid channel info. This makes hard to use
this cli. Changed to keep iterate after hits the invalid channel
info.
Joshua Colp [Tue, 19 Feb 2019 16:06:32 +0000 (16:06 +0000)]
CI: Use tmpfs option to Docker instead of mount.
Some tests require Asterisk to execute scripts which
are stored in /tmp. When mount is used for tmpfs there
is no ability to allow scripts to be executed from
that location.
This change switches to using tmpfs which can be told
to allow executables to be run from /tmp.
Torrey Searle [Tue, 12 Feb 2019 09:50:55 +0000 (10:50 +0100)]
res/res_rtp_asterisk: clear smoother when local bridging
p2p_write updates txformat but doesn't require a smoother. If a smoother
was created by another bridge type the smoother could fall out of date causing
one way audio issues. To prevent this the smoother is now destroyed on the
start of native bridge.
Joshua Colp [Thu, 7 Feb 2019 15:52:56 +0000 (15:52 +0000)]
ci: Rerun unit tests when non-code changes occur.
This change makes it so that even if non-code changes
occur (such as commit message changing) unit tests
will still be run and result in a verification.
Kevin Harwell [Thu, 31 Jan 2019 19:29:05 +0000 (13:29 -0600)]
res_pjsip_registrar: lock transport monitor when setting 'removing' flag
A previous patch attempt to mitigate blocked threads on transport shutdown for
a given contact. It was thought that a second lock could be avoided by checking
the 'removing' flag on the transport monitor twice (once before and once after
the normal named aor locking). However as with usual threading issues if the
timing was right the original problem still occured.
This patch adds locking around the first 'removing' flag check and set, thus
nullifying the secondary check, so it was removed.
Joshua Colp [Wed, 6 Feb 2019 12:16:01 +0000 (12:16 +0000)]
res_odbc: Add basic query logging.
When Asterisk is connected and used with a database the response
time of the database can cause problems in Asterisk if it is long.
Normally the only way to see this problem would be to retrieve a
backtrace from Asterisk and examine where things are blocked, or
examine the database to see if there is any indication of a
problem.
This change adds some basic query logging to make it easier to
investigate such a problem. When logging is enabled res_odbc will
now keep track of the number of queries executed, as well as the
query that has taken the longest time to execute. There is also
an option which will cause a WARNING message to be output if a
query takes longer than a configurable amount of time to execute.
This makes it easier and clearer for users that their database may
be experiencing a problem that could impact Asterisk.
George Joseph [Mon, 4 Feb 2019 13:09:57 +0000 (06:09 -0700)]
bundled-jansson: On OpenSuse Leap libjansson.a was placed in lib64
On OpenSuse Leap, libjansson.a is installed in
third-party/jansson/dest/lib64 instead of lib (which is where
the top-level makeopts looks). This causes a link failure.
* Updated jansson/Makefile to add an explicit --libdir to force
the installation to third-party/jansson/dest/lib.
Ben Ford [Tue, 29 Jan 2019 16:48:49 +0000 (10:48 -0600)]
res_stasis: Auto-create context and extens on Stasis app launch.
At AstriCon, there was a strong desire for the ability to completely
bypass dialplan when using ARI. This is possible through the automatic
creation of a context and a couple of extensions whenever an application
is started.
For example, if you have an application named 'ari-example', a context
named 'stasis-ari-example' will be automatically created whenever this
application is started as long as one does not already exist. Two
extensions (a match-all extension for Stasis and a 'h' extension) are
created within this context. Any endpoint that registers to Asterisk
within this context will send all calls to the corresponding Stasis
application. When the application is destroyed, the context is removed.
Kevin Harwell [Tue, 15 Jan 2019 23:20:30 +0000 (17:20 -0600)]
pjsip/config_global: regcontext context not created
The context specified by 'regcontext' was not being created, so when Asterisk
attempted to later dynamically add an extension it would fail. This patch now
creates the context if a 'regcontext' is specified.
George Joseph [Tue, 22 Jan 2019 15:02:06 +0000 (08:02 -0700)]
media_index.c: Refactored so it doesn't cache the index
Testing revealed that the cache added no benefit but that it could
consume excessive memory.
Two new index related functions were created:
ast_sounds_get_index_for_file() and ast_media_index_update_for_file()
which restrict index updating to specific sound files.
The original ast_sounds_get_index() and ast_media_index_update()
calls are still available but since they no longer cache the results
internally, developers should re-use an index they may already have
instead of calling ast_sounds_get_index() repeatedly. If information
for only a single file is needed, ast_sounds_get_index_for_file()
should be called instead of ast_sounds_get_index().
The media_index directory scan code was elimininated in favor of
using the existing ast_file_read_dirs() function.
Since there's no more cache, ast_sounds_index_init now only
registers the sounds cli commands instead of generating the
initial index and subscribing to stasis format register/unregister
messages.
ast_sounds_reindex() is now a no-op but left for backwards
compatibility.
loader.c no longer registers "sounds" as a special reload target.
Both the sounds cli commands and the sounds ari resources were
refactored to only call ast_sounds_get_index() once per invocation
and to use ast_sounds_get_index_for_file() when a specific sound
file is requested.
Fix deadlock handling subscribe req during res_parking reload
Split destroy_hint method to separate hint removal and extension hint
state changed callback, the latter now called via stasis.
This avoids deadlock between res_parking reload that is removing the
parking lot and the related hint and subscribe requests coming for the
same parking lot.
eyalhasson [Tue, 22 Jan 2019 15:24:23 +0000 (17:24 +0200)]
format_g726: add support for seeking
Added support for the seek function in format_g726
so playback can start from anywhere.
Before the fix, playback of g726 files
always started from the beginning.
Paulo Vicentini [Thu, 8 Nov 2018 10:21:03 +0000 (11:21 +0100)]
res/res_pjsip: Fix crash due to misuse of session->media between threads.
This patch makes sure that thread running ast_taskprocessor_execute
cannot suddenly dispose the session->media object making the other
threads (running pbx_thread / bridge_channel_ind_thread) crash when they
try to access the pointer to invalid memory. We were experiencing a crash due
to a misuse of session->media container between threads running
(bridge_channel_ind_thread/pbx_thread) and the thread running
ast_taskprocessor_execute. Depending on the SIP flow (during a disconnection)
and the threads' code path, the session->media container was being destroyed
(and set to NULL) by the thread running ast_taskprocessor_execute while the
thread running t38_framehook_read was still referring to it.
Now res_pjsip_t38 is referring a session_media in a datastore.
Jeremy Lainé [Wed, 23 Jan 2019 10:45:56 +0000 (11:45 +0100)]
res_http_websocket: ensure control frames do not interfere with data
Control frames (PING / PONG / CLOSE) can be received in the middle of a
fragmented message. In order to ensure they do not interfere with the
reassembly buffer, we exit early and do not return the payload to the
caller.
projects / thirdparty / asterisk.git / log
