core/buildsystem: check the actual compiler being version
Make compiler check use the output of the actual compiler being
used as reported by the CC variable, instead of unconditionally
running the "gcc" binary. Also only run the check if the compiler
is gcc or a cross-compile gcc.
Checks the PJSIP global setting value.
If it is true (default) it adds the norefersub capability to PJSIP.
If it is false (disabled) it does not add the norefersub capability
to PJSIP.
This is useful for Cisco switches that do not follow RFC4488.
ASTERISK-28375 #close Reported-by: Dan Cropp
Change-Id: I0b1c28ebc905d881f4a16e752715487a688b30e9
George Joseph [Fri, 12 Apr 2019 16:32:44 +0000 (10:32 -0600)]
CI: Move test group config files to Jenkins
One of the downaides of having things like test configuration
in the git repo is that it can't be changed at runtime. You have
to create a review for the changes and merge it mefore it will
take effect.
This review moves the data currently held in
tests/CI/periodic-dailyTestGroups.json and
tests/CI/gateTestGroups.json into a Jenkins Config File attached
to the job definitions. This allows us to alter it from the
Jenkins UI at runtime. The original files stay in the repo
as documentation.
Sean Bright [Tue, 9 Apr 2019 15:10:12 +0000 (11:10 -0400)]
app_voicemail: Cleanup stale lock files on module load
If Asterisk crashes while a VM directory is locked, lock files in the VM
spool directory will not get properly cleaned up. We now clear them on
module load.
Sean Bright [Thu, 11 Apr 2019 20:48:49 +0000 (16:48 -0400)]
res_ael: Create consistent label names across reloads
Reset the internal counter that the AEL2 compiler uses for unique label
names before compiling. This keeps dialplan labels consistent across
reloads assuming the AEL2 has not changed.
Sean Bright [Thu, 11 Apr 2019 16:03:07 +0000 (12:03 -0400)]
res_ael: Fix pattern matching against literal '+'
When generating the regular expression that matches against existing
extensions, we need to escape literal characters that can also be
regular expression metacharacters. This was already being done for '*'
but we need to do the same for '+'.
In passing, remove some unreachable code - strcmp() is already run
immediately when entering extension_matches().
Ben Ford [Mon, 1 Apr 2019 19:57:14 +0000 (14:57 -0500)]
build: Revise CHANGES and UPGRADE.txt handling.
This changes the way that we handle adding changes to CHANGES and
UPGRADE.txt. The reason for this is because whenever someone needed to
make a change to one of these files and someone else had already done
so, you would run into merge conflicts. With this new setup, there will
never be merge conflicts since all changes will be documented in the
doc/<file>-staging directory. The release script is now responsible for
merging all of these changes into the appropriate files.
There is a special format that these files have to follow in order to be
parsed. The files do not need to have a meaningful name, but it is
strongly recommended. For example, if you made a change to pjsip, you
may have something like this "res_pjsip_relative_title", where
"relative_title" is something more descriptive than that. Inside each
file, you will need a subject line for your change, followed by a
description. There can be multiple subject lines. The file may look
something like this:
Subject: res_pjsip
Subject: Core
A description that explains the changes made and why. The release
script will handle the bulleting and section separators!
You can still separate with new lines within your
description.
The headers ("Subject" and "Master-Only") are case sensative, but the
value for "Master-Only" ("true" or "True") is not.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
When extconfig.conf file is parsed, the code previously searched for
character comma without verifying if error (null or blank). This caused
a segmentation error.
sungtae kim [Thu, 21 Mar 2019 23:09:14 +0000 (00:09 +0100)]
main/json.c: Added app_name, app_data to channel type
It was difficult to check the channel's current application and
parameters using ARI for current channels. Added app_name, app_data
items to show the current application information.
Asterisk assumes that dlopen() will always run the constructor of a
shared library and every dlclose() will run its destructor. But dlopen()
may be permanent, meaning the constructor will only be run once, as is
the case with musl libc.
With a permanent dlopen() the Asterisk module loader does not work
correctly, because it's expectations regarding when the constructors and
destructors are run are not met. In fact a segmentation fault will occur
when the first module is "re-opened" that has AST_MODFLAG_GLOBAL_SYMBOLS
set (the dlopen() does not call the constructor, resource_being_loaded
is not set to NULL, then strlen is called with NULL instead of a string,
see issue ASTERISK-28319).
This commit adds code to the loader that will manually run the
constructors/destructors of the (non-builtin) modules where needed. To
achieve this a new ao2 container (linked list) is started and filled
with objects that contain the names of the modules and the pointers to
their respective info structs.
This behavior can be activated when configuring Asterisk
(--enable-permanent-dlopen). By default this is disabled, of course.
ASTERISK-28319 #close
Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
Change-Id: I86693a0ecf25d5ba81c73773a03df4abc3426875
Ben Ford [Wed, 3 Apr 2019 15:24:06 +0000 (10:24 -0500)]
build: Fix compiler warnings/errors.
The compiler complained about a couple of variables that weren't
initialized but were being used. Initializing them to NULL resolves the
warnings/errors.
Kevin Harwell [Wed, 27 Mar 2019 17:59:30 +0000 (12:59 -0500)]
bridge_softmix: use a float type to store the internal REMB bitrate
REMB's exponent is 6-bits (0..63) and has a mantissa of 18-bits. We were using
an unsigned integer to represent the bitrate. However, that type is not large
enough to hold all potential bitrate values. If the bitrate is large enough
bits were being shifted off the "front" of the mantissa, which caused the
wrong value to be sent to the browser.
This patch makes it so it now uses a float type to hold the bitrate. Using a
float allows for all bitrate values to be correctly represented.
res/res_rtp_asterisk: Enable rxjitter calculation for video
It looks like we're not properly calculating jitter values on received
video streams. This patch enables the code that does jitter calculations
for those streams.
Ben Ford [Tue, 26 Mar 2019 19:56:37 +0000 (14:56 -0500)]
alembic: Fix errors during upgrade head.
When trying to upgrade using alembic, a couple different errors kept
popping up that prevented the upgrade. An additional parameter was
needed when changing the schema for mwi_subscribe_replaces_unsolicited
from an integer to an enum. When changing from a string to an enum, the
type needed to be cast for postgresql. The other issue was a parameter
being used during column creation that did not exist.
After fixing the upgrade process, it revealed errors with the downgrade
process. One was a variable not being defined in the downgrade function,
and the other was tables not existing when using MySQL. This was due to
a context check that should have encompassed MySQL, but in the end was
not doing so.
Ben Ford [Wed, 27 Mar 2019 16:03:40 +0000 (11:03 -0500)]
build: Add staging directories for future changes.
This is the first step in changing the release process so that changes
made to the CHANGES and UPGRADE.txt files do not result in merge
conflicts every time multiple people modify these files. The changes
made will go in these new directories: doc/CHANGES-staging and
doc/UPGRADE-staging. The README.md files explain how things will work,
but here's a little overview. When you make a change that would go in
either CHANGES or UPGRADE.txt, this should instead be documented in a
new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory,
respectively. The format will look like this:
Subject: res_pjsip
A description that explains the changes made and why. The release
script will handle the bulleting and section separators! The
'Subject:' header is case-sensitive.
You can still separate with new lines within your description.
Subject: res_ari
Master-Only: true
You can have more than one subject, and they don't have to be the
same! Also, the 'Master-Only' header should always be true and is
also case-sensitive (but the value is not - you can have 'true' or
'True'). This header will only ever be present in the master branch.
For more information, check out the wiki page:
https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt
This is an initial change for ASTERISK_28111. Functionally, this will
make no difference, but it will prep the directories for when the
changes from CHANGES and UPGRADE.txt are extracted.
Joshua Colp [Mon, 25 Mar 2019 11:34:09 +0000 (08:34 -0300)]
manager: Use separate lock for session event notification.
When notifying a manager session that new events were available
the same lock was used that was also held when doing things within
the session (such as sending events out). If the manager session
blocked for a period of time this would cause a back up of messages
in Stasis and would also block any other sessions from receiving
events.
This change adds a separate lock to the manager session which is
strictly used for notifying it that new events are available.
sungtae kim [Wed, 6 Mar 2019 22:21:42 +0000 (23:21 +0100)]
res/res_ari: Added timestamp as a requirement for all ARI events
Changed to requirement to having timestamp for all of ARI events.
The below ARI events were changed to having timestamp.
PlaybackStarted, PlaybackContinuing, PlaybackFinished,
RecordingStarted, RecordingFinished, RecordingFailed,
ApplicationReplaced, ApplicationMoveFailed
Alexei Gradinari [Mon, 25 Mar 2019 16:42:27 +0000 (12:42 -0400)]
res_config_odbc: set empty extended field as a single whitespace
If Realtime @ variable value is NULL or empty or contains only whitespaces
then when we try to retrieve it using PJSIP_ENDPOINT we get WARNING
pjsip_endpoint_function_read: Unknown property @my_var for PJSIP endpoint.
And the variable is missing in the result of CLI pjsip show endpoint.
main/taskprocessor: Increase max name length of taskprocessors
Since the new names went in, the maximum taskprocessor name is too
short. This patch increases the name field to a length to better
handle the new names.
George Joseph [Thu, 14 Mar 2019 16:46:53 +0000 (10:46 -0600)]
sorcery.c: Sorcery enhancements for wizard management
Added ability to specifiy a wizard is read-only when applying
it to a specific object type. This allows you to specify
create, update and delete callbacks for the wizard but limit
which object types can use them.
Added the ability to allow an object type to have multiple
wizards of the same type. This is indicated when a wizard
is added to a specific object type.
Added 3 new sorcery wizard functions:
* ast_sorcery_object_type_insert_wizard which does the same thing
as the existing ast_sorcery_insert_wizard_mapping function but
accepts the new read-only and allot-duplicates flags and also
returns the ast_sorcery_wizard structure used and it's internal
data structure. This allows immediate use of the wizard's
callbacks without having to register a "wizard mapped" observer.
* ast_sorcery_object_type_apply_wizard which does the same
thing as the existing ast_sorcery_apply_wizard_mapping function
but has the added capabilities of
ast_sorcery_object_type_insert_wizard.
* ast_sorcery_object_type_remove_wizard which removes a wizard
matching both its name and its original argument string.
* The original logic in __ast_sorcery_insert_wizard_mapping was moved
to __ast_sorcery_object_type_insert_wizard and enhanced for the
new capabilities, then __ast_sorcery_insert_wizard_mapping was
refactored to just call __ast_sorcery_insert_wizard_mapping.
* Added a unit test to test_sorcery.c to test the read-only
capability.
sungtae kim [Sun, 10 Mar 2019 22:53:36 +0000 (23:53 +0100)]
res/res_stasis: Fixed wrong StasisEnd timestamp
Because StasisEnd's timestamp created it's own timestamp, it makes
wrong timestamp, compare to other channel event(ChannelDestroyed).
Fixed to getting a timestamp from the Channel's timestamp.
Sean Bright [Thu, 14 Mar 2019 14:55:30 +0000 (10:55 -0400)]
vector: Add AST_VECTOR_COMPACT() to reclaim wasted space
This might be useful in situations where you are loading an undetermined number
of items into a vector and don't want to keep (potentially) 2x the necessary
memory around indefinitely.
George Joseph [Fri, 8 Mar 2019 15:40:38 +0000 (08:40 -0700)]
app.c: Remove deletion of pool topic on mwi state delete
As part of an earlier voicemail refactor, ast_delete_mwi_state_full
was modified to remove the pool topic for a mailbox when the state
was deleted. This was an attempt to prevent stale topics from
accumulating when app_voicemail was reloaded and a mailbox went
away. Unfortunately because of the fact that when app_voicemail
reloads, ALL mailboxes are deleted then only current ones recreated,
topics were being removed from the pool that still had subscribers
on them, then recreated as new topics of the same name. So now
modules like res_pjsip_mwi are listening on a topic that will
never receive any messages because app_voicemail is publishing on
a different topic that happens to have the same name. The solutiuon
to this is not easy and given that accumulating topics for
deleted mailboxes is less evil that not sending NOTIFYs...
* Removed the call to stasis_topic_pool_delete_topic in
ast_delete_mwi_state_full.
Also:
* Fixed a topic reference leak in res_pjsip_mwi
mwi_stasis_subscription_alloc.
* Added some debugging to mwi_stasis_subscription_alloc,
stasis_topic_create, and topic_dtor.
* Fixed a topic reference leak in an error path in
internal_stasis_subscribe.
ASTERISK-28306 Reported-by: Jared Hull
Change-Id: Id7da0990b3ac4be4b58491536b35f41291247b27
cirillor [Sat, 9 Mar 2019 14:39:26 +0000 (11:39 -0300)]
Variable ALTCONF ignored when service is used in Debian
When variable ALTCONF is defined, the command start prints the message
"Unable to open specified master config file '"/etc/asterisk/asteris..."
and use default configurations.
Kevin Harwell [Tue, 12 Mar 2019 20:54:23 +0000 (15:54 -0500)]
AMI/ARI: Bump non-breaking version numbers
Increasing the non-breaking AMI and ARI version numbers due to changes and
additions in those API's. Note, some changes may be forthcoming (will be added
between now and the next release of Asterisk), thus not listed here. As well
a few changes listed below may have been released in a previous release of
Asterisk, but the API version numbers were not increased at that time, so
including here.
AMI:
* app_queue: set the wrapuptime from AddQueueMember application - e806990
* res_pjsip: option for ContactStatus event updates - 7f22c9f
ARI:
* bridging: Add creation timestamps - 0d70120
* res_stasis: Add ability to switch applications - 50a4b61
* ARI event type filtering - 1c5def4
* Added ARI resource /ari/asterisk/ping - 19fc99a
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.
partial-inlining: disable partial-inlining if gcc>=8.2.1
Apply flag -fno-partial-inlining on default optimization if and only if
gcc version >= 8.2.1 (this is the current ver on Fedora and Ubuntu).
This is done to avoid a bug that causes arithmetic calculations to fail
if the following conditions are met:
1. TEST_FRAMEWORK on
2. DONT_OPTIMIZE off
3. Fedora and Ubuntu
4. GCC 8.2.1
5. There must exist a certain combination of multithreading.
6. Optimization level -O2 and -O3
7. Flag -fpartial-inline activated (default when optimization level>=2)
The following link points to a similar gcc bug reported in 2015. This leads me
to believe the bug has regressed. Note I am not able to replicate this bug
in an environment other than Asterisk + Test Framework + Test_cel because the
multithreading combination that causes it seems to be unique. Therefore I
am temporarily abandoning any thoughts of reporting the new occurrence of this
bug to gcc.gnu.org. https://gcc.gnu.org/bugzilla/show_bug.cgi?id=65307
Joshua Colp [Thu, 7 Mar 2019 12:28:31 +0000 (08:28 -0400)]
stasis: Improve topic/subscription names and statistics.
Topic names now follow: <subsystem>:<functionality>[/<object>]
This ensures that they are all unique, and also provides better
insight in to what each topic is for.
Subscriber ids now also use the main topic name they are
subscribed to and an incrementing integer as their identifier to
make it easier to understand what the subscription is primarily
responsible for.
Both the CLI commands for listing topic and subscription statistics
now sort to make it a bit easier to see what is going on.
Subscriptions will now show all topics that they are receiving messages
from, not just the main topic they were subscribed to.
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.
projects / thirdparty / asterisk.git / log
