Certain channel options are not set anywhere or
exposed in any way to users, making them unusable.
This exposes some of these options which make sense
for users to manipulate at runtime.
Sean Bright [Tue, 7 Nov 2023 20:03:53 +0000 (15:03 -0500)]
app.c: Allow ampersands in playback lists to be escaped.
Any function or application that accepts a `&`-separated list of
filenames can now include a literal `&` in a filename by wrapping the
entire filename in single quotes, e.g.:
UpgradeNote: Ampersands in URLs passed to the `Playback()`,
`Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or
`Queue()` applications as filename arguments can now be escaped by
single quoting the filename. Additionally, this is also possible when
using the `CONFBRIDGE` dialplan function, or configuring various
features in `confbridge.conf` and `queues.conf`.
Sean Bright [Thu, 9 Nov 2023 14:11:33 +0000 (09:11 -0500)]
live_ast: Add astcachedir to generated asterisk.conf.
`astcachedir` (added in b0842713) was not added to `live_ast` so
continued to point to the system `/var/cache` directory instead of the
one in the live environment.
app_followme.c: Grab reference on nativeformats before using it
Fixes a crash due to a lack of proper reference on the nativeformats
object before passing it into ast_request(). Also found potentially
similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c
Naveen Albert [Thu, 9 Nov 2023 15:12:15 +0000 (10:12 -0500)]
configs: Improve documentation for bandwidth in iax.conf.
This improves the documentation for the bandwidth setting
in iax.conf by making it clearer what the ramifications
of this setting are. It also changes the sample default
from low to high, since only high is compatible with good
codecs that people will want to use in the vast majority
of cases, and this is a common gotcha that trips up new users.
Naveen Albert [Wed, 9 Aug 2023 22:12:55 +0000 (22:12 +0000)]
logger: Add channel-based filtering.
This adds the ability to filter console
logging by channel or groups of channels.
This can be useful on busy systems where
an administrator would like to analyze certain
calls in detail. A dialplan function is also
included for the purpose of assigning a channel
to a group (e.g. by tenant, or some other metric).
ASTERISK-30483 #close
Resolves: #242
UserNote: The console log can now be filtered by
channels or groups of channels, using the
logger filter CLI commands.
George Joseph [Tue, 7 Nov 2023 23:24:37 +0000 (16:24 -0700)]
codec_ilbc: Disable system ilbc if version >= 3.0.0
Fedora 37 started shipping ilbc 3.0.4 which we don't yet support.
configure.ac now checks the system for "libilbc < 3" instead of
just "libilbc". If true, the system version of ilbc will be used.
If not, the version included at codecs/ilbc will be used.
George Joseph [Tue, 31 Oct 2023 21:08:14 +0000 (15:08 -0600)]
chan_pjsip: Add PJSIPHangup dialplan app and manager action
See UserNote below.
Exposed the existing Hangup AMI action in manager.c so we can use
all of it's channel search and AMI protocol handling without
duplicating that code in dialplan_functions.c.
Added a lookup function to res_pjsip.c that takes in the
string represenation of the pjsip_status_code enum and returns
the actual status code. I.E. ast_sip_str2rc("DECLINE") returns
603. This allows the caller to specify PJSIPHangup(decline) in
the dialplan, just like Hangup(call_rejected).
Also extracted the XML documentation to its own file since it was
almost as large as the code itself.
UserNote: A new dialplan app PJSIPHangup and AMI action allows you
to hang up an unanswered incoming PJSIP call with a specific SIP
response code in the 400 -> 699 range.
Sean Bright [Mon, 6 Nov 2023 16:45:33 +0000 (11:45 -0500)]
chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE
in a frame was one that may not have any data - such as the CALLTOKEN
IE in an NEW request - it was not getting displayed.
Naveen Albert [Wed, 27 Sep 2023 12:49:07 +0000 (08:49 -0400)]
app_directory: Add ADSI support to Directory.
This adds optional ADSI support to the Directory
application, which allows callers with ADSI CPE
to navigate the Directory system significantly
faster than is possible using the audio prompts.
Callers can see the directory name (and optionally
extension) on their screenphone and confirm or
reject a match immediately rather than waiting
for it to be spelled out, enhancing usability.
Naveen Albert [Wed, 9 Aug 2023 22:57:41 +0000 (22:57 +0000)]
core_local: Fix local channel parsing with slashes.
Currently, trying to call a Local channel with a slash
in the extension will fail due to the parsing of characters
after such a slash as being dial modifiers. Additionally,
core_local is inconsistent and incomplete with
its parsing of Local dial strings in that sometimes it
uses the first slash and at other times it uses the last.
For instance, something like DAHDI/5 or PJSIP/device
is a perfectly usable extension in the dialplan, but Local
channels in particular prevent these from being called.
This creates inconsistent behavior for users, since using
a slash in an extension is perfectly acceptable, and using
a Goto to accomplish this works fine, but if specified
through a Local channel, the parsing prevents this.
This fixes this by explicitly parsing options from the
last slash in the extension, rather than the first one,
which doesn't cause an issue for extensions with slashes.
Naveen Albert [Mon, 30 Oct 2023 13:27:20 +0000 (09:27 -0400)]
app_voicemail: Add AMI event for mailbox PIN changes.
This adds an AMI event that is emitted whenever a
mailbox password is successfully changed, allowing
AMI consumers to process these.
UserNote: The VoicemailPasswordChange event is
now emitted whenever a mailbox password is updated,
containing the mailbox information and the new
password.
In simple_bridge_join, we were sending topology change requests
even when the new and old topologies were the same. In some
circumstances, this can cause unnecessary re-invites and even
a re-invite flood. We now suppress those.
Mike Bradeen [Thu, 7 Sep 2023 17:21:32 +0000 (11:21 -0600)]
res_speech: allow speech to translate input channel
* Allow res_speech to translate the input channel if the
format is translatable to a format suppored by the
speech provider.
Resolves: #129
UserNote: res_speech now supports translation of an input channel
to a format supported by the speech provider, provided a translation
path is available between the source format and provider capabilites.
George Joseph [Thu, 19 Oct 2023 13:40:26 +0000 (07:40 -0600)]
res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
Commit f66f77f last year prevents the res_pjsip_exten_state and
res_pjsip_mwi modules from unloading due to possible pjproject
asserts if the modules are reloaded. A side effect of the
implementation is that the taskprocessors these modules use aren't
being released. When asterisk is doing a graceful shutdown, it
waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all
taskprocessors to stop but since those 2 modules don't release
theirs, the shutdown hangs for that amount of time.
This change allows the modules to be unloaded and their resources to
be released when ast_shutdown_final is true.
sungtae kim [Fri, 22 Sep 2023 17:32:43 +0000 (02:32 +0900)]
res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
This commit introduces an extension to the endpoint and relevant
resource sizes for PJSIP, transitioning from its current 40-character
constraint to a more versatile 255-character capacity. This enhancement
significantly overcomes limitations related to domain qualification and
practical usage, ultimately delivering improved functionality. In
addition, it includes adjustments to accommodate the expanded realm size
within the ARI, specifically enhancing the maximum realm length.
Resolves: #345
UserNote: With this update, the PJSIP realm lengths have been extended
to support up to 255 characters.
UpgradeNote: As part of this update, the maximum allowable length
for PJSIP endpoints and relevant resources has been increased from
40 to 255 characters. To take advantage of this enhancement, it is
recommended to run the necessary procedures (e.g., Alembic) to
update your schemas.
George Joseph [Tue, 17 Oct 2023 12:59:43 +0000 (06:59 -0600)]
.github: New PR Submit workflows
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.
George Joseph [Tue, 17 Oct 2023 12:59:43 +0000 (06:59 -0600)]
.github: New PR Submit workflows
The workflows that get triggered when PRs are submitted or updated
have been replaced with ones that are more secure and have
a higher level of parallelism.
Mike Bradeen [Mon, 2 Oct 2023 18:35:32 +0000 (12:35 -0600)]
res_stasis: signal when new command is queued
res_statsis's app loop sleeps for up to .2s waiting on input
to a channel before re-checking the command queue. This can
cause delays between channel setup and bridge.
This change is to send a SIGURG on the sleeping thread when
a new command is enqueued. This exits the sleeping thread out
of the ast_waitfor() call triggering the new command being
processed on the channel immediately.
Resolves: #362
UserNote: Call setup times should be significantly improved
when using ARI.
George Joseph [Thu, 5 Oct 2023 14:18:17 +0000 (08:18 -0600)]
logger.h: Add ability to change the prefix on SCOPE_TRACE output
You can now define the _TRACE_PREFIX_ macro to change the
default trace line prefix of "file:line function" to
something else. Full documentation in logger.h.
George Joseph [Thu, 21 Sep 2023 17:12:31 +0000 (11:12 -0600)]
Add libjwt to third-party
The current STIR/SHAKEN implementation is not currently usable due
to encryption issues. Rather than trying to futz with OpenSSL and
the the current code, we can take advantage of the existing
capabilities of libjwt but we first need to add it to the
third-party infrastructure already in place for jansson and
pjproject.
A few tweaks were also made to the third-party infrastructure as
a whole. The jansson "dest" install directory was renamed "dist"
to better match convention, and the third-party Makefile was updated
to clean all product directories not just the ones currently in
use.
Mike Bradeen [Tue, 26 Sep 2023 19:19:27 +0000 (13:19 -0600)]
res_pjsip: update qualify_timeout documentation with DNS note
The documentation on qualify_timeout does not explicitly state that the timeout
includes any time required to perform any needed DNS queries on the endpoint.
If the OPTIONS response is delayed due to the DNS query, it can still render an
endpoint as Unreachable if the net time is enough for qualify_timeout to expire.
Naveen Albert [Mon, 4 Sep 2023 11:58:23 +0000 (07:58 -0400)]
chan_dahdi: Clarify scope of callgroup/pickupgroup.
Internally, chan_dahdi only applies callgroup and
pickupgroup to FXO signalled channels, but this is
not documented anywhere. This is now documented in
the sample config, and a warning is emitted if a
user tries configuring these settings for channel
types that do not support these settings, since they
will not have any effect.
Mike Bradeen [Thu, 21 Sep 2023 18:54:45 +0000 (12:54 -0600)]
res_speech_aeap: add aeap error handling
res_speech_aeap previously did not register an error handler
with aeap, so it was not notified of a disconnect. This resulted
in SpeechBackground never exiting upon a websocket disconnect.
Naveen Albert [Wed, 27 Sep 2023 12:28:50 +0000 (08:28 -0400)]
app_voicemail: Disable ADSI if unavailable.
If ADSI is available on a channel, app_voicemail will repeatedly
try to use ADSI, even if there is no CPE that supports it. This
leads to many unnecessary delays during the session. If ADSI is
available but ADSI setup fails, we now disable it to prevent
further attempts to use ADSI during the session.
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.
George Joseph [Wed, 13 Sep 2023 15:18:04 +0000 (09:18 -0600)]
lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
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.
George Joseph [Fri, 15 Sep 2023 17:19:18 +0000 (11:19 -0600)]
asterisk.c: Use the euid's home directory to read/write cli history
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.
res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
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.
Naveen Albert [Sat, 9 Sep 2023 15:24:37 +0000 (11:24 -0400)]
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 5b8fea93d106332bc0faa4b7fa8a6ea71e546cac
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.
George Joseph [Mon, 11 Sep 2023 14:27:41 +0000 (08:27 -0600)]
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.
George Joseph [Fri, 15 Sep 2023 16:17:11 +0000 (10:17 -0600)]
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 "".
George Joseph [Wed, 13 Sep 2023 20:08:02 +0000 (14:08 -0600)]
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.
George Joseph [Mon, 11 Sep 2023 16:19:41 +0000 (10:19 -0600)]
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.
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.
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.
George Joseph [Fri, 15 Dec 2023 16:37:54 +0000 (09:37 -0700)]
res_rtp_asterisk: Fix regression issues with DTLS client check
* Since ICE candidates are used for the check and pjproject is
required to use ICE, res_rtp_asterisk was failing to compile
when pjproject wasn't available. The check is now wrapped
with an #ifdef HAVE_PJPROJECT.
* The rtp->ice_active_remote_candidates container was being
used to check the address on incoming packets but that
container doesn't contain peer reflexive candidates discovered
during negotiation. This was causing the check to fail
where it shouldn't. We now check against pjproject's
real_ice->rcand array which will contain those candidates.
* Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where
we weren't zeroing out sin->sin_zero before returning. This
was causing ast_sockaddr_cmp() to always return false when
one of the inputs was converted from a pj_sockaddr, even
if both inputs had the same address and port.
res_pjsip_header_funcs: Duplicate new header value, don't copy.
When updating an existing header the 'update' code incorrectly
just copied the new value into the existing buffer. If the
new value exceeded the available buffer size memory outside
of the buffer would be written into, potentially causing
a crash.
This change makes it so that the 'update' now duplicates
the new header value instead of copying it into the existing
buffer.
Mike Bradeen [Tue, 25 Jul 2023 16:23:12 +0000 (10:23 -0600)]
res_pjsip: disable raw bad packet logging
Add patch to split the log level for invalid packets received on the
signaling port. The warning regarding the packet will move to level 2
so that it can still be displayed, while the raw packet will be at level
4.
George Joseph [Thu, 9 Nov 2023 16:14:03 +0000 (09:14 -0700)]
res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
When ICE is in use, we can prevent a possible DOS attack by allowing
DTLS protocol messages (client hello, etc) only from sources that
are in the active remote candidates list.
Ben Ford [Mon, 13 Nov 2023 17:08:50 +0000 (11:08 -0600)]
manager.c: Prevent path traversal with GetConfig.
When using AMI GetConfig, it was possible to access files outside of the
Asterisk configuration directory by using filenames with ".." and "./"
even while live_dangerously was not enabled. This change resolves the
full path and ensures we are still in the configuration directory before
attempting to access the file.
George Joseph [Tue, 5 Sep 2023 15:51:19 +0000 (09:51 -0600)]
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.
Bastian Triller [Mon, 28 Aug 2023 07:16:43 +0000 (09:16 +0200)]
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.
Naveen Albert [Mon, 28 Aug 2023 17:51:23 +0000 (13:51 -0400)]
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.
George Joseph [Mon, 28 Aug 2023 13:11:27 +0000 (07:11 -0600)]
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).
George Joseph [Fri, 18 Aug 2023 16:20:47 +0000 (10:20 -0600)]
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.
Naveen Albert [Wed, 9 Aug 2023 22:04:23 +0000 (22:04 +0000)]
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.
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.
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.
MikeNaso [Tue, 8 Aug 2023 14:23:37 +0000 (10:23 -0400)]
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.
projects / thirdparty / asterisk.git / log
