Commit Graph

34110 Commits

Author SHA1 Message Date
Naveen Albert
3b027d1e47 chan_console: Fix deadlock caused by unclean thread exit.
To terminate a console channel, stop_stream causes pthread_cancel
to make stream_monitor exit. However, commit 5b8fea93d1
added locking to this function which results in deadlock due to
the stream_monitor thread being killed while it's holding the pvt lock.

To resolve this, a flag is now set and read to indicate abort, so
the use of pthread_cancel and pthread_kill can be avoided altogether.

Resolves: #308
2023-09-20 19:15:45 +00:00
George Joseph
0e0f99db1d file.c: Add ability to search custom dir for sounds
To better co-exist with sounds files that may be managed by
packages, custom sound files may now be placed in
AST_DATA_DIR/sounds/custom instead of the standard
AST_DATA_DIR/sounds/<lang> directory.  If the new
"sounds_search_custom_dir" option in asterisk.conf is set
to "true", asterisk will search the custom directory for sounds
files before searching the standard directory.  For performance
reasons, the "sounds_search_custom_dir" defaults to "false".

Resolves: #315

UserNote: A new option "sounds_search_custom_dir" has been added to
asterisk.conf that allows asterisk to search
AST_DATA_DIR/sounds/custom for sounds files before searching the
standard AST_DATA_DIR/sounds/<lang> directory.
2023-09-20 19:14:58 +00:00
Naveen Albert
f93138bcad chan_iax2: Improve authentication debugging.
Improves and adds some logging to make it easier
for users to debug authentication issues.

Resolves: #286
2023-09-20 15:38:36 +00:00
Vitezslav Novy
56244a7371 res_rtp_asterisk: fix wrong counter management in ioqueue objects
In function  rtp_ioqueue_thread_remove counter in ioqueue object is not decreased
which prevents unused ICE TURN threads from being removed.

Resolves: #301
2023-09-20 15:03:07 +00:00
George Joseph
71d75373f9 res_pjsip_pubsub: Add body_type to test_handler for unit tests
The ast_sip_subscription_handler "test_handler" used for the unit
tests didn't set "body_type" so the NULL value was causing
a SEGV in build_subscription_tree().  It's now set to "".

Resolves: #335
2023-09-20 12:29:01 +00:00
George Joseph
42f82d55b0 make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
The previous behavior of make_buildopts_h was to not add the
non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS,
REF_DEBUG, etc. to the buildopts.h file because "it caused
ccache to invalidate files and extended compile times". They're
only defined by passing them on the gcc command line with '-D'
options.   In practice, including them in the include file rarely
causes any impact because the only time ccache cares is if you
actually change an option so the hit occurrs only once after
you change it.

OK so why would we want to include them?  Many IDEs follow the
include files to resolve defines and if the options aren't in an
include file, it can cause the IDE to mark blocks of "ifdeffed"
code as unused when they're really not.

So...

* Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H
  which tells make_buildopts_h to include the non-ABI-breaking
  flags in buildopts.h as well as the ABI-breaking ones. The default
  is disabled to preserve current behavior.  As before though,
  only the ABI-breaking flags appear in AST_BUILDOPTS and only
  those are used to calculate AST_BUILDOPT_SUM.
  A new AST_BUILDOPT_ALL define was created to capture all of the
  flags.

* make_version_c was streamlined to use buildopts.h and also to
  create asterisk_build_opts_all[] and ast_get_build_opts_all(void)

* "core show settings" now shows both AST_BUILDOPTS and
  AST_BUILDOPTS_ALL.

UserNote: The "Build Options" entry in the "core show settings"
CLI command has been renamed to "ABI related Build Options" and
a new entry named "All Build Options" has been added that shows
both breaking and non-breaking options.
2023-09-14 17:58:17 +00:00
Mike Bradeen
2265eafb3a func_periodic_hook: Add hangup step to avoid timeout
func_periodic_hook does not hangup after playback, relying on hangup
which keeps the channel alive longer than necessary.

Resolves: #325
2023-09-13 17:36:37 +00:00
Sean Bright
b952a8c38d res_stasis_recording.c: Save recording state when unmuted.
Fixes #322
2023-09-13 17:34:34 +00:00
Mike Bradeen
3759d034cc res_speech_aeap: check for null format on response
* Fixed issue in res_speech_aeap when unable to provide an
  input format to check against.
2023-09-13 17:33:36 +00:00
George Joseph
cff637a24f func_periodic_hook: Don't truncate channel name
func_periodic_hook was truncating long channel names which
causes issues when you need to run other dialplan functions/apps
on the channel.

Resolves: #319
2023-09-13 15:17:51 +00:00
George Joseph
6bd9f9ae81 safe_asterisk: Change directory permissions to 755
If the safe_asterisk script detects that the /var/lib/asterisk
directory doesn't exist, it now creates it with 755 permissions
instead of 770.  safe_asterisk needing to create that directory
should be extremely rare though because it's normally created
by 'make install' which already sets the permissions to 755.

Resolves: #316
2023-09-13 15:17:11 +00:00
Maximilian Fridrich
1af2ae177c chan_rtp: Implement RTP glue for UnicastRTP channels
Resolves: #298

UserNote: The dial string option 'g' was added to the UnicastRTP channel
which enables RTP glue and therefore native RTP bridges with those
channels.
2023-09-07 11:29:27 +00:00
Jaco Kroon
130c3ab792 app_queue: periodic announcement configurable start time.
This newly introduced periodic-announce-startdelay makes it possible to
configure the initial start delay of the first periodic announcement
after which periodic-announce-frequency takes over.

UserNote: Introduce a new queue configuration option called
'periodic-announce-startdelay' which will vary the normal (historic)
behavior of starting the periodic announcement cycle at
periodic-announce-frequency seconds after entering the queue to start
the periodic announcement cycle at period-announce-startdelay seconds
after joining the queue.  The default behavior if this config option is
not set remains unchanged.

Signed-off-by: Jaco Kroon <jaco@uls.co.za>
2023-09-07 11:28:30 +00:00
Joshua C. Colp
182ad6926b variables: Add additional variable dialplan functions.
Using the Set dialplan application does not actually
delete channel or global variables. Instead the
variables are set to an empty value.

This change adds two dialplan functions,
GLOBAL_DELETE and DELETE which can be used to
delete global and channel variables instead
of just setting them to empty.

There is also no ability within the dialplan to
determine if a global or channel variable has
actually been set or not.

This change also adds two dialplan functions,
GLOBAL_EXISTS and VARIABLE_EXISTS which can be
used to determine if a global or channel variable
has been set or not.

Resolves: #289

UserNote: Four new dialplan functions have been added.
GLOBAL_DELETE and DELETE have been added which allows
the deletion of global and channel variables.
GLOBAL_EXISTS and VARIABLE_EXISTS have been added
which checks whether a global or channel variable has
been set.
2023-09-07 11:27:53 +00:00
George Joseph
c457784bd0 ari-stubs: Fix more local anchor references
Also allow CreateDocs job to be run manually with default branches.
2023-09-05 13:36:20 -06:00
George Joseph
b68006f3a1 ari-stubs: Fix more local anchor references
Also allow CreateDocs job to be run manually with default branches.
2023-09-05 13:05:49 -06:00
George Joseph
c4508accc6 ari-stubs: Fix broken documentation anchors
All of the links that reference page anchors with capital letters in
the ids (#Something) have been changed to lower case to match the
anchors that are generated by mkdocs.
2023-09-05 09:55:52 -06:00
Bastian Triller
9284dca636 res_pjsip_session: Send Session Interval too small response
Handle session interval lower than endpoint's configured minimum timer
when sending first answer. Timer setting is checked during this step and
needs to handled appropriately.
Before this change, no response was sent at all. After this change a
response with 422 Session Interval too small is sent to UAC.
2023-08-31 14:22:23 +00:00
George Joseph
83964ede1f .github: Update workflow-application-token-action to v2 2023-08-31 07:26:09 -06:00
Naveen Albert
c032d38d3f app_dial: Fix infinite loop when sending digits.
If the called party hangs up while digits are being
sent, -1 is returned to indicate so, but app_dial
was not checking the return value, resulting in
the hangup being lost and looping forever until
the caller manually hangs up the channel. We now
abort if digit sending fails.

ASTERISK-29428 #close

Resolves: #281
2023-08-31 13:20:06 +00:00
Mike Bradeen
e9b269692f app_voicemail: Fix for loop declarations
Resolve for loop initial declarations added in cli changes.

Resolves: #275
2023-08-30 13:05:16 +00:00
George Joseph
f8438714ca alembic: Fix quoting of the 100rel column
Add quoting around the ps_endpoints 100rel column in the ALTER
statements.  Although alembic doesn't complain when generating
sql statements, postgresql does (rightly so).

Resolves: #274
2023-08-29 11:10:08 +00:00
Naveen Albert
9e6266e008 pbx.c: Fix gcc 12 compiler warning.
Resolves: #277
2023-08-28 13:38:03 +00:00
zhengsh
eab40e755e app_audiosocket: Fixed timeout with -1 to avoid busy loop.
Resolves: asterisk#234
2023-08-28 13:36:49 +00:00
George Joseph
644ad2f9cd download_externals: Fix a few version related issues
* Fixed issue with the script not parsing the new tag format for
  certified releases.  The format changed from certified/18.9-cert5
  to certified-18.9-cert5.

* Fixed issue where the asterisk version wasn't being considered
  when looking for cached versions.

Resolves: #263
2023-08-22 13:32:10 +00:00
Maximilian Fridrich
48e6a482c2 main/refer.c: Fix double free in refer_data_destructor + potential leak
Resolves: #267
2023-08-22 13:30:51 +00:00
Naveen Albert
cd0bfe193f sig_analog: Add Called Subscriber Held capability.
This adds support for Called Subscriber Held for FXS
lines, which allows users to go on hook when receiving
a call and resume the call later from another phone on
the same line, without disconnecting the call. This is
a convenience mechanism that most real PSTN telephone
switches support.

ASTERISK-30372 #close

Resolves: #240

UserNote: Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station  user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
2023-08-22 13:30:26 +00:00
Matthew Fredrickson
1d7ae8d227 Revert "app_stack: Print proper exit location for PBXless channels."
This reverts commit 617dad4cba.

apps/app_stack.c: Revert buggy gosub patch

This seems to break the case when a predial macro calls a gosub.
When the gosub calls return, the Return function outputs:

app_stack.c:423 return_exec: Return without Gosub: stack is empty

This returns -1 to the calling macro, which returns to app_dial
and causes the call to hangup instead of proceeding with the macro
that invoked the gosub.

Resolves: #253
2023-08-16 14:45:22 +00:00
Jason D. McCormick
2f02095da8 install_prereq: Fix dependency install on aarch64.
Fixes dependency solutions in install_prereq for Debian aarch64
platforms. install_prereq was attempting to forcibly install 32-bit
armhf packages due to the aptitude search for dependencies.

Resolves: #37
2023-08-14 17:26:30 +00:00
MikeNaso
720813dc97 res_pjsip.c: Set contact_user on incoming call local Contact header
If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.

Resolves: #226
2023-08-14 17:21:35 +00:00
Sean Bright
1171beb7e4 extconfig: Allow explicit DB result set ordering to be disabled.
Added a new boolean configuration flag -
`order_multi_row_results_by_initial_column` - to both res_pgsql.conf
and res_config_odbc.conf that allows the administrator to disable the
explicit `ORDER BY` that was previously being added to all generated
SQL statements that returned multiple rows.

Fixes: #179
2023-08-14 17:20:05 +00:00
George Joseph
21b0522abd rest-api: Run make ari-stubs
An earlier cherry-pick that involved rest-api somehow didn't include
a comment change in res/ari/resource_endpoints.h.  This commit
corrects that.  No changes other than the comment.
2023-08-14 17:19:07 +00:00
Naveen Albert
2179082eaf res_pjsip_header_funcs: Make prefix argument optional.
The documentation for PJSIP_HEADERS claims that
prefix is optional, but in the code it is actually not.
However, there is no inherent reason for this, as users
may want to retrieve all header names, not just those
beginning with a certain prefix.

This makes the prefix optional for this function,
simply fetching all header names if not specified.
As a result, the documentation is now correct.

Resolves: #230

UserNote: The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
2023-08-14 17:18:10 +00:00
George Joseph
d3ee0e6516 pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
The default is 32 with 8 being used by pjproject itself.  Recent
commits have put us over the limit resulting in assertions in
pjproject.  Since this value is used in invites, dialogs,
transports and subscriptions as well as the global pjproject
endpoint, we don't want to increase it too much.

Resolves: #255
2023-08-14 17:17:22 +00:00
Joshua C. Colp
a75035be55 manager: Tolerate stasis messages with no channel snapshot.
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
2023-08-11 13:28:58 +00:00
George Joseph
ea7e719d71 Remove unneeded CHANGES and UPGRADE files 2023-08-09 18:58:48 +00:00
George Joseph
68349125a8 Update version for Asterisk 21 21.0.0-pre1 2023-08-09 18:58:48 +00:00
Maximilian Fridrich
51a7b18038 core/ari/pjsip: Add refer mechanism
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.

Resolves: #71

UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
2023-08-09 15:10:46 +00:00
Naveen Albert
d16046e41f chan_dahdi: Allow autoreoriginating after hangup.
Currently, if an FXS channel is still off hook when
all calls on the line have hung up, the user is provided
reorder tone until going back on hook again.

In addition to not reflecting what most commercial switches
actually do, it's very common for switches to automatically
reoriginate for the user so that dial tone is provided without
the user having to depress and release the hookswitch manually.
This can increase convenience for users.

This behavior is now supported for kewlstart FXS channels.
It's supported only for kewlstart (FXOKS) mainly because the
behavior doesn't make any sense for ground start channels,
and loop start signalling doesn't provide the necessary DAHDI
event that makes this easy to implement. Likely almost everyone
is using FXOKS over FXOLS anyways since FXOLS is pretty useless
these days.

ASTERISK-30357 #close

Resolves: #224

UserNote: The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
2023-08-09 14:51:35 +00:00
Joshua C. Colp
806515597e audiohook: Unlock channel in mute if no audiohooks present.
In the case where mute was called on a channel that had no
audiohooks the code was not unlocking the channel, resulting
in a deadlock.

Resolves: #233
2023-08-09 14:50:07 +00:00
Naveen Albert
e1a1ae933b sig_analog: Allow three-way flash to time out to silence.
sig_analog allows users to flash and use the three-way dial
tone as a primitive hold function, simply by never timing
it out.

Some systems allow this dial tone to time out to silence,
so the user is not annoyed by a persistent dial tone.
This option allows the dial tone to time out normally to
silence.

ASTERISK-30004 #close
Resolves: #205

UserNote: The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
2023-08-04 14:31:18 +00:00
Holger Hans Peter Freyther
f335da6b74 res_prometheus: Do not generate broken metrics
In 8d6fdf9c3a invisible bridges were
skipped but that lead to producing metrics with no name and no help.

Keep track of the number of metrics configured and then only emit these.
Add a basic testcase that verifies that there is no '(NULL)' in the
output.

ASTERISK-30474
2023-08-04 14:21:23 +00:00
Sean Bright
c52b4ce11c res_pjsip: Enable TLS v1.3 if present.
Fixes #221

UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by
the underlying PJSIP library. The bundled version of PJSIP supports
TLS v1.3.
2023-08-04 14:20:56 +00:00
phoneben
83dd36ba13 func_cut: Add example to documentation.
This adds an example to the XML documentation clarifying usage
of the CUT function to address a common misusage.
2023-08-04 10:57:13 +00:00
Sean Bright
5c4cbeff87 extensions.conf.sample: Remove reference to missing context.
c3ff4648 removed the [iaxtel700] context but neglected to remove
references to it.

This commit addresses that and also removes iaxtel and freeworlddialup
references from other config files.
2023-07-21 17:49:34 +00:00
Sean Bright
b22aabe64a func_export: Use correct function argument as variable name.
Fixes #208
2023-07-18 14:55:39 +00:00
Joshua C. Colp
6b6880072b app_queue: Add support for applying caller priority change immediately.
The app_queue module provides both an AMI action and a CLI command
to change the priority of a caller in a queue. Up to now this change
of priority has only been reflected to new callers into the queue.

This change adds an "immediate" option to both the AMI action and
CLI command which immediately applies the priority change respective
to the other callers already in the queue. This can allow, for example,
a caller to be placed at the head of the queue immediately if their
priority is sufficient.

Resolves: #202

UserNote: The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.
2023-07-18 13:03:06 +00:00
George Joseph
d0e5f2f6be .github: Fix cherry-pick reminder issues 2023-07-17 09:23:08 -06:00
George Joseph
8a864bcdba app.h: Move declaration of ast_getdata_result before its first use
The ast_app_getdata() and ast_app_getdata_terminator() declarations
in app.h were changed recently to return enum ast_getdata_result
(which is how they were defined in app.c).  The existing
declaration of ast_getdata_result in app.h was about 1000 lines
after those functions however so under certain circumstances,
a "use before declaration" error was thrown by the compiler.
The declaration of the enum was therefore moved to before those
functions.

Resolves: #200
2023-07-12 17:44:50 +00:00
Sean Bright
508657879e chan_iax2.c: Avoid crash with IAX2 switch support.
A change made in 82cebaa0 did not properly handle the case when a
channel was not provided, triggering a crash. ast_check_hangup(...)
does not protect against NULL pointers.

Fixes #180
2023-07-12 17:40:07 +00:00