Some providers require a multiple of 20 for the maxptime or fail to complete calls,
e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used.
Resolves: #260
Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS.
Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead
to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just
causes the lock calls to loop over trylock in 200us intervals until
the lock is obtained and spits out log messages if it takes more
than 5 seconds. From a code perspective, the only reason they were
tied together was for logging. So... The ifdefs in lock.c were
refactored to allow DETECT_DEADLOCKS to be enabled without
also enabling DEBUG_THREADS.
Resolves: #321
UserNote: You no longer need to select DEBUG_THREADS to use
DETECT_DEADLOCKS. This removes a significant amount of overhead
if you just want to detect possible deadlocks vs needing full
lock tracing.
The CLI .asterisk_history file is read from/written to the directory
specified by the HOME environment variable. If the root user starts
asterisk with the -U/-G options, or with runuser/rungroup set in
asterisk.conf, the asterisk process is started as root but then it
calls setuid/setgid to set the new user/group. This does NOT reset
the HOME environment variable to the new user's home directory
though so it's still left as "/root". In this case, the new user
will almost certainly NOT have access to read from or write to the
history file.
* Added function process_histfile() which calls
getpwuid(geteuid()) and uses pw->dir as the home directory
instead of the HOME environment variable.
* ast_el_read_default_histfile() and ast_el_write_default_histfile()
have been modified to use the new process_histfile()
function.
Resolves: #337
From the gdb information, ast_websocket_read reads a message successfully,
then transport_read is called in the serializer. During execution of pjsip_transport_down,
ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop.
After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages.
This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue.
In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop.
Resolves: asterisk#299
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
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.
In function rtp_ioqueue_thread_remove counter in ioqueue object is not decreased
which prevents unused ICE TURN threads from being removed.
Resolves: #301
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.
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
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
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.
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>
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.
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.
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.
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
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
* 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
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.
app_macro sometimes would crash due to datastore list corruption on the
channel because of lack of locking around find and create process for
the macro datastore. This patch locks the channel lock prior to protect
against this problem.
Resolves: #265
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
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
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
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
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.
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.
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
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.
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.
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.
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.
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
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.
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.
