The original trigger for setting the RTP stats in ast_softhangup() came from
an ARI issue where stats weren't being set in time to be reported on STASIS_END
events. The thought was that setting them in a common place like ast_softhangup()
would ensure the stats were set in possibly other scenarios. Unfortunately,
setting the RTP stats variables in ast_softhangup() broke ABI as it required
that no channel locks be held which was not the case earlier.
Given that the original issue was ARI, we can move setting the stats to
ast_ari_channels_hangup() in resource_channels just before it calls
ast_softhangup(). This might not catch all cases of the stats not being set,
but it won't break ABI or deadlock either.
Resolves: #1928
If a hostname is specified for stunaddr in rtp.conf, periodic DNS resolution
is enabled based on the TTL returned in the DNS results. If the TTL returned
is 0, it means that the next time the IP address is needed, it must be
looked up again. I.E. Don't cache. Historically (and incorrectly) however,
res_rtp_asterisk stopped the periodic resolution and never re-resolved the
hostname again.
Besides what's mentioned in the user notes...
* Additional debugging was added in various STUN/DNS functions.
* The `rtp show settings` CLI command shows more detailed STUN info.
* Some debugging was added to dns_core.c and dns_recurring.c.
UserNote: A new `stunaddr_reresolve_ttl_0` parameter has been added to rtp.conf
that allows control over what happens when a STUN server hostname lookup
returns a TTL of 0. The values can be set as follows:
- 'no': This is the historical (and current default) behavior of not doing
any further lookups and continuing to use the last successful result until
Asterisk is restarted or rtp.conf is reloaded.
- 'yes': Use the last cached result for the current call but trigger
re-resolution in the background for the benefit of future calls.
If the result of the background lookup is a ttl > 0, periodic resolution
will be restarted otherwise the next call will use the new cached value
and will trigger a background lookup again.
UserNote: A new CLI command `rtp resolve stun hostname` has been added
that will force a resolution of the STUN hostname and (re)start periodic
resolution if the result has a TTL > 0.
Resolves: #1858
Rather than merely showing
dtls_verify : Yes/No
in pjsip show endpoint xxx it will now be shown what exactly is being
checked, ie, one of:
dtls_verify : No
dtls_verify : Fingerprint
dtls_verify : Certificate
dtls_verify : Yes
Where Yes implies both Fingerprint and Certificate.
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
If we are sending digits (either DTMF, MF, or SF) to the called channel
after receiving progress or a wink, and the callee hangs up before we
have finished sending it digits, there are several problems that can ensue:
* If the callee hung up without answering, the calling channel would
hang up and not continue in the dialplan.
* If the callee *did* answer before hanging up, the answer was never
passed through to the caller, since this gets "eaten" by the various
digit streaming functions and is never processed by app_dial.
This is generally an edge case that occurs due to some kind of signaling
failure, but to better handle this:
* Set to_answer to 0 to prevent hangup on the exit path, just like other
parts of wait_for_answer.
* Better document this usage of to_answer.
* If the channel did answer while it was receiving digits, manually
answer the calling channel before we abort. The call would not continue
in the dialplan anyways (either before or after this fix), but technically
the call was answered, so the CDRs should probably reflect that, and this
mirrors the behavior of calls which normally do not continue.
Resolves: #1915
UserNote: If a called channel sends progress or wink and the caller begins
sending digits but the callee answers and then hangs up before digit
sending can finish, the call is now answered before being disconnected.
If the callee hangs up without answering, the call now continues in
the dialplan.
When a message is sent via ARI, the ARI endpoint only provides a To
field which is also used as destination field. This means that the To
field might not necessarily contain a SIP URI but might instead specify
an Asterisk endpoint (in MessageDestinationInfo format). This led to
many warnings even though the message was sent correctly.
The fix is to only call `ast_sip_update_to_uri` if the To field starts
with the sip: or sips: scheme.
Resolves: #1357
crypto_utils uses ast_asprintf to allocate the search string when checking the
certificate subject, but was not using ast_free to free it. This caused a crash
when Asterisk was built with malloc_debug
Resolves: #1921
Due to stasis filtering the stasis callback for AMI type messages is
guaranteed to only receive messages that can be turned into AMI events,
so remove the check done in the callback.
The sessions container usage for the stasis callbacks has also been
simplified by having a reference on the message router subscription
instead of having to acquire the sessions from the global object each
time.
Modified the bridge playback teardown so the worker thread removes only the
playback control, while the after-bridge callback removes the playback
wrapper once the announcer has actually left the bridge.
This avoids a stale window where a new playback request could create a
replacement announcer before the old announcer had fully exited the holding
bridge.
Also replaced the flexible trailing bridge_id storage in the shared worker
thread data with an optional bridge_id pointer, since recording paths use the
same structure without a bridge id.
Fixes: #1861
ast_softhangup() was locking the channel before calling ast_rtp_instance_set_stats_vars()
which, if the channel was in a bridge, then locked the bridge peer channel. If another
thread attempted to set bridge variables on the peer, it would lock that channel first,
then this channel causing a lock inversion. ast_softhangup() now holds the channel lock
while retrieving the rtp instance, then unlocks it before calling
ast_rtp_instance_set_stats_vars(), then locks it again after it returns.
Resolves: #1907
When a PJSIP endpoint is configured with set_var invoking a dialplan
function (e.g. PJSIP_HEADER(add,...)), chan_pjsip_new() calls
pbx_builtin_setvar_helper() while holding the channel lock.
For function-style variables, this dispatches to ast_func_write()
which, in the case of PJSIP_HEADER, calls
ast_sip_push_task_wait_serializer() -- blocking synchronously while
the channel lock is held.
If a concurrent operation (ARI, AMI, rtp_check_timeout) traverses
the channels container via ast_channel_get_by_name(), it acquires
the container lock then tries to lock individual channels in the
iteration callback (by_uniqueid_cb/by_name_cb). When the serializer
thread also needs the container lock, a circular dependency forms:
channel_lock -> serializer_wait -> container_lock -> channel_lock
This causes a complete Asterisk freeze. In the observed case, 36
threads were blocked on the container lock until res_freeze_check
triggered SIGABRT after its 30-second timeout.
Unlock the channel before iterating endpoint channel_vars so that
dialplan functions can block without holding the channel lock. Re-lock
the channel for ast_channel_stage_snapshot_done() so the batched
snapshot is published under lock and captures the full channel state
including the variables set during the loop.
Fixes: #1872
Add rtp_port_start and rtp_port_end options to PJSIP endpoint
configuration, allowing each endpoint to use a dedicated RTP port
range instead of the global rtp.conf setting.
This is useful for scenarios where different endpoints need isolated
port ranges, such as firewall rules per trunk, multi-tenant systems,
or network QoS policies tied to port ranges.
The implementation adds ast_rtp_instance_new_with_port_range() to the
RTP engine API, which sets the port range on the instance before the
engine allocates the transport. The default RTP engine
(res_rtp_asterisk) checks for per-instance overrides in
rtp_allocate_transport() and falls back to the global range when
none is set.
Both options must be set together, with values >= 1024 and
rtp_port_end > rtp_port_start. Setting both to 0 (the default)
preserves existing behavior.
Resolves: https://github.com/asterisk/asterisk-feature-requests/issues/71
UserNote: PJSIP endpoints now support rtp_port_start and
rtp_port_end options to configure a dedicated RTP port range per
endpoint, overriding the global rtp.conf setting.
UpgradeNote: An alembic database migration has been added to add
the rtp_port_start and rtp_port_end columns to the ps_endpoints
table. Run "alembic upgrade head" to apply the schema change.
DeveloperNote: New public API: ast_rtp_instance_new_with_port_range()
creates an RTP instance with a per-instance port range.
ast_rtp_instance_get_port_start() and ast_rtp_instance_get_port_end()
allow RTP engines to query the override. Third-party RTP engines can
use these getters to support per-instance port ranges.
app_queue: Fix raise_respect_min not copied in copy_rules() causing rN rules to be ignored.
`copy_rules()` never copied `raise_respect_min` into the per-call rule list, so the flag was always 0 when a timed penaltychange rule fired, making `rN` behave like plain `N` and raising members below `min_penalty` that should have been excluded.
Also fixes `update_qe_rule()` not propagating the flag from `qe->pr` to `qe`, and dropping the `r` prefix when saving back to `QUEUE_RAISE_PENALTY`.
Resolves: #1901
app_voicemail_odbc: fix msgnum race and crash on failed STORE
Two concurrent callers leaving voicemail to the same mailbox could be
assigned the same msgnum because ast_unlock_path() was called before
STORE(), allowing a second thread to read the same LAST_MSG_INDEX()
before the first INSERT committed. The losing thread got a duplicate
key error, but execution continued into notify_new_message() ->
RETRIEVE() because the STORE() return value was not checked.
RETRIEVE() then fetched the winning thread's DB row, mmap'd its blob
size against the locally truncated file, and crashed with SIGBUS.
Hold the path lock through STORE() and bail out on failure.
Fixes: #1653
1. session_cleanup() now saves the websocket type before unlinking the
session from the session registry. This prevents a FRACK when cleaning
up per-call websockets when MALLOC_DEBUG is used.
2. session_shutdown_cb() and outbound_sessions_load() now call
pthread_cancel() to cancel the session handler thread to prevent the
thread from continually trying to connect to a server after the
connection config has been removed by a reload. This required the
thread to use pthread_cleanup_push() to clean up its reference to the
session instead of RAII because RAII destructors don't get run when
pthread_cancel() is used.
Resolves: #1894
`__STDC_VERSION__` is specific to C but up until gcc 16, the g++ compiler
also defined it. With g++ 16.0 it's no longer defined (which is the correct
behavior) so compiling channelstorage_cpp_map_name_id.cc fails. The
check for `__STDC_VERSION__` in compat.h is now skipped if we're compiling
a C++ source file.
Resolves: #1903
Backport pjsip/pjproject#4941 which fixes a build/link failure when
compiling against OpenSSL < 1.1.0 (e.g. OpenSSL 1.0.2k on CentOS 7).
Two symbols introduced in OpenSSL 1.1.x were called unconditionally
in ssl_sock_ossl.c without version guards:
- `TLS_method()` in `init_ossl_ctx()` is now guarded with
`OPENSSL_VERSION_NUMBER < 0x10100000L`, falling back to
`SSLv23_method()` on older OpenSSL.
- `SSL_CTX_set_ciphersuites()` is now guarded with
`OPENSSL_VERSION_NUMBER >= 0x1010100fL` since this function
was introduced in OpenSSL 1.1.1 and is absent in 1.0.x.
Without this fix, linking fails with:
undefined reference to `TLS_method'
undefined reference to `SSL_CTX_set_ciphersuites'
when building Asterisk with bundled pjproject on systems such as
CentOS 7 with OpenSSL 1.0.2k.
Resolves: #1892
Line 2729 has `#if HAVE_LIBEDIT_IS_UNICODE` instead if `#ifdef`. Since
macros defined by autoconf are either set to `1` or not set at all,
older distros where libedit isn't unicode won't have that macro defined
and will fail to compile.
Resolves: #1896
cdrel_custom: fix SQLite compatibility for versions < 3.20.0
Replace sqlite3_prepare_v3 + SQLITE_PREPARE_PERSISTENT with a version-guarded fallback to sqlite3_prepare_v2 for older SQLite builds.
Resolves: #1885
The default (codec) still uses the codec provided samples. Additionally
different sample_types can be used with eg: `translate sampletype speech`
and then running `core show translation comp 10` to measure performance
of different audio scenarios.
Resolves: #1807
Adds two optional modules:
res_stasis_broadcast.so: Infrastructure for broadcasting a single incoming
channel to multiple ARI applications with atomic first-claim-wins semantics.
app_stasis_broadcast.so: Provides the StasisBroadcast() dialplan application
which invokes the broadcast infrastructure.
Both modules are self-contained; if neither is loaded there is zero runtime
impact. Loading them does not alter existing Stasis or ARI behavior unless
explicitly used.
Key Features (only active when modules are loaded):
Fisher-Yates shuffled broadcast dispatch for fair claim races
Atomic claim operations using mutex + condition variable signaling
Configurable broadcast timeouts
Safe regex application filtering with validation to mitigate ReDoS risk
Thread-safe channel variable snapshotting (channel locked during reads)
Late-claim safety: broadcast context kept alive until after the Stasis
session ends so concurrent claimants always receive 409 Conflict rather
than 404 Not Found
Memory safety via RAII_VAR, ast_json_ref/unref, and ao2 reference counting
Components Added:
res/res_stasis_broadcast.c: Core broadcast + claim logic
apps/app_stasis_broadcast.c: StasisBroadcast() dialplan application
include/asterisk/stasis_app_broadcast.h: Public API header
res/ari/resource_events.c: Integrates POST /ari/events/claim endpoint
rest-api/api-docs/events.json: New CallBroadcast and CallClaimed events
Implementation Notes:
Broadcast contexts reside in an ao2 hash container keyed by channel id. Each
context holds atomic claim state, winner application name, timeout metadata,
and a condition variable for waiters. Broadcast contexts are kept alive until
after stasis_app_exec() returns so that concurrent claimants racing against
the timeout always receive 409 Conflict. Broadcast dispatch calls
stasis_app_send() directly for each matching application in shuffled order.
Regex filters are validated with bounded length, group depth, quantified
group count, and alternation limits to reduce pathological backtracking.
Timeout calculation uses timespec arithmetic with overflow-safe millisecond
remainder handling. Event JSON follows existing Stasis/ARI conventions;
references are managed correctly to avoid leaks or double frees.
Optional Nature / Impact:
No changes to existing APIs, events, or applications when absent.
Clean fallback: systems ignoring the modules behave identically to prior
versions.
Development was assisted by Claude (Anthropic). All generated code has been
reviewed, tested, and is understood by the author.
UserNote: New optional modules res_stasis_broadcast.so and
app_stasis_broadcast.so enable broadcasting an incoming channel to multiple
ARI applications. The first application to successfully claim (via
POST /ari/events/claim) wins channel control. StasisBroadcast() dialplan
application initiates broadcasts. CallBroadcast and CallClaimed events notify
applications. When modules are not loaded, behavior is unchanged.
DeveloperNote: New public APIs in stasis_app_broadcast.h:
stasis_app_broadcast_channel(), stasis_app_claim_channel(),
stasis_app_broadcast_winner(), and stasis_app_broadcast_wait(). New ARI event
types (CallBroadcast, CallClaimed) added to events.json. All code is isolated;
no existing ABI modified.
This commit implements the handling of non-voice or DTMF frames like the
chan_websocket handling added in #1588. Rather than treating unsupported
frames as fatal errors, silently ignore CNG frames and log a warning for
other unsupported types.
Before ast_func_read and ast_func_write call their respective read and write
callbacks for registered dialplan functions, they use the module pointer in
the registered ast_custom_function structure to increment the module use
count. They then decrement the usecount when the callback returns. This
prevents the providing module from being unloaded while there's a call using
the function.
Some modules, notably func_odbc, create and destroy dialplan functions based
on the contents of a config file. Since the ast_custom_function structure is
dynamically allocated, it could be destroyed on reload which means when the
module's read or write callback returns to the ast_func calls it would try to
decrement the usecount using the module pointer from an ast_custom_function
structure that has already been freed. Proper locking or reference counting
by the module can reduce the possibility of this happening but it can't
prevent it because it doesn't have control after its read or write callback
has returned to ast_func_read or ast_func_write.
To address this, ast_func_read, ast_func_read2 and ast_func_write save the
module pointer to a local variable before calling the module's callback,
then use the saved pointer to decrement the use count. The module
pointer will always be valid if the module is loaded regardless of the
state of the ast_custom_function structure.
Resolves: #1818
Add a property to the CHANNEL method to retrieve the auth method,
which can be used to retrieve the specific auth method actually
negotiated for a call (e.g. RSA, MD5, etc.).
Also clean up some of the documentation for the secure properties
to clarify how these relate to call encryption.
Resolves: #1878
UserNote: CHANNEL(auth_method) can now be used to retrieve the
auth method negotiated for a call on IAX2 channels.
pjproject 2.16 (bundled) fails to build on GCC 4.8 (CentOS/RHEL 7)
due to a false positive C11 atomics detection introduced in pjproject
commit #4570. A fix has been submitted upstream to pjproject (#4933).
Adding a local patch to third-party/pjproject/patches/ until a fixed
version of pjproject is bundled in Asterisk.
Fixes build error:
../src/pj/os_core_unix.c:52:27: fatal error: stdatomic.h: No such file or directory
Resolves: #1883
The rtp_ioqueue_thread_destroy() function was destroying the the ioqueue
thread and releasing its pool but not destroying the ioqueue itself. This
was causing the ioqueue's epoll file descriptor to leak.
Resolves: #1867
Introduces res_pjsip_maintenance, a loadable module that allows
operators to place individual PJSIP endpoints into maintenance mode
at runtime without unregistering or disabling them.
While an endpoint is in maintenance mode:
* New inbound INVITE and SUBSCRIBE dialogs are rejected with
503 Service Unavailable and a Retry-After: 300 header.
* In-progress dialogs (re-INVITE, UPDATE, BYE, etc.) are
unaffected and complete normally.
* Outbound originations via Dial() or ARI originate are refused
before any SIP session is created.
State is held in-memory only and is cleared on module unload
or Asterisk restart.
This module was developed with AI assistance (Claude). All code
has been reviewed and tested by the author, who takes full
responsibility for the submission.
CLI interface:
pjsip set maintenance <on|off> <endpoint|all>
pjsip show maintenance [endpoint]
AMI interface:
Action: PJSIPSetMaintenance
Endpoint: <name>|all
State: on|off
Action: PJSIPShowMaintenance
Endpoint: <name> (optional; omit to list all)
Emits PJSIPMaintenanceStatus events per result, followed by
PJSIPMaintenanceStatusComplete. State changes also emit an
unsolicited PJSIPMaintenanceStatus event.
To support outbound blocking, a new session_create callback is
added to ast_sip_session_supplement. Supplements that set this
callback are invoked at the start of ast_sip_session_create_outgoing()
in res_pjsip_session, before any dialog or invite session resources
are allocated. res_pjsip_maintenance registers itself as a session
supplement and uses this callback to gate outbound session creation
on a per-endpoint basis.
MODULEINFO:
<depend>pjproject</depend>
<depend>res_pjsip</depend>
<depend>res_pjsip_session</depend>
UserNote: New module res_pjsip_maintenance adds runtime maintenance
mode for PJSIP endpoints. Use "pjsip set maintenance <on|off>
<endpoint|all>" to enable or disable, and "pjsip show maintenance"
to list affected endpoints. AMI actions PJSIPSetMaintenance and
PJSIPShowMaintenance provide programmatic access. No configuration
file changes required.
DeveloperNote: ast_sip_session_supplement gains a new optional
callback - int (*session_create)(struct ast_sip_endpoint *endpoint,
const char *destination). It is called from the global supplement
list (not per-session) at the start of ast_sip_session_create_outgoing()
via ast_sip_session_check_supplement_create(). Returning non-zero
blocks the outgoing session. Modules that need to gate outbound
SIP session creation should register a supplement with this callback
set rather than hooking into chan_pjsip directly.
Commit 2da221e217 added a missing abort
if datalen < 0 check on a code path and an assertion inside
iax_frame_wrap if we ever encountered a frame with a negative frame
length (which will eventually cause a crash).
Add another missing abort check for negative datalen, exposed by this
assertion. (Similar to the previous commit, this is a video frame with
a datalen of -1).
Resolves: #1865
The PJSIP outbound registration API has undocumented behavior when reconfiguring
the outbound registration if the expires value being set is the same as what was
previously set.
In this case PJSIP will remove the Expires header entirely from subsequent
outbound REGISTER requests. To eliminate this as an issue we now check the current
expires value against the configured expires value and only apply it if it differs.
This ensures that outbound REGISTER requests always contain an Expires header.
Resolves: #1859
When res_pjsip is reloaded directly, it does the sorcery reload in a pjsip
servant thread as it's supposed to. res_pjsip_config_wizard however
was not which was leading to occasional deadlocks. It now does the reload
in a servant thread just like res_pjsip.
Resolves: #1855
The pjsua Python module and the pjsua/pjsystest apps were used by the
Asterisk Test Suite for SIP simulation in dev mode builds. They are now
fully obsolete for three independent reasons:
1. **pjsua Python bindings officially deprecated upstream.** The pjproject
maintainers added `pjsip-apps/src/python/DEPRECATED.txt` directing
users to the PJSUA2 SWIG binding instead. A build-fix PR
(https://github.com/pjsip/pjproject/pull/4892) was closed by the
maintainer explicitly citing this deprecation.
2. **Removed from the Asterisk Test Suite.** As confirmed by @mbradeen:
> *"We had to get rid of pjsua when we went to Python3 because it would
> hang due to a conflict between async calls within pjsua and twisted.
> There are still some old references to tests we couldn't fully convert
> to sipp, but those are skipped."*
3. **Broken and unmaintained.** Building with Python 2.7 (the only version
`configure.ac` searched for) fails with:
```
_pjsua.c: error: 'INIT_RETURN' undeclared (first use in this function)
```
due to a bug in pjproject 2.16's `python3_compat.h` that upstream
declined to fix.
This PR removes all pjsua-related build artifacts from Asterisk's bundled
pjproject build: the pjsua and pjsystest application binaries, the deprecated
Python (`_pjsua.so`) bindings, the `asterisk_malloc_debug.c` stubs, and the
`PYTHONDEV` detection from `configure.ac`. Also removes `libpjsua` from
Asterisk's main linker flags.
DeveloperNote: The pjsua and pjsystest application binaries, the deprecated
Python pjsua bindings (`_pjsua.so`), and the `asterisk_malloc_debug.c` stub
implementations are no longer built or installed as part of the bundled
pjproject dev mode build. The `PYTHONDEV` (python2.7-dev) build dependency
is also removed. Developers who relied on the pjsua binary for Test Suite
SIP simulation should use SIPp instead, which is the current Asterisk Test
Suite standard.
Fixes: #1840
When the res_cdrel_custom modules is reloaded and the config has not been changed asterisk should not free the old config. Otherwise the connection to the database will be closed and no new connection will be opened.
Resolves: #1852
Several issues are resolved:
* Internally, floats were used for timestamp values but this could result
in wrapping so they've been changed to doubles.
* Historically, the default CEL eventtime format is `<seconds>.<microseconds>`
with `<microseconds>` always being 6 digits. This should have continued to be
the case but res_cdrel_custom wasn't checking the `dateformat` setting in
cel.conf and was defaulting to `%F %T`. res_cdrel_custom now gets the default
date format from cel.conf, which will be whatever the `dateformat` parameter
is set to or `<seconds>.<microseconds>` if not set.
* The timeval field formatter for both CDR and CEL wasn't handling custom
strftime format strings correctly. This is now fixed so you should be able
to specifiy custom strftime format strings for the CEL `eventtime` and CDR
`start`, `answer` and `end` fields. For example: `eventtime(%FT%T%z)`.
Resolves: #1844Resolves: #1845
Change rawdata buffer from char to unsigned char to prevent
sign-extension of TLV length bytes >= 0x80. On signed-char
platforms (all Asterisk builds due to -fsigned-char in
configure.ac), these values become negative when assigned to
int, bypass the `if (res > 32)` bounds check, and reach
memcpy as size_t producing a ~18 EB read that immediately
crashes with SIGSEGV.
Affects DAHDI analog (FXO) channels only. Not reachable
via SIP, PRI/BRI, or DTMF-based Caller ID.
Fixes: #1839
ast_channel_datastore_find() and ast_channel_datastore_add() must only be
called while the channel is locked (see channel.h). raise_exception() and the
EXCEPTION dialplan function read path accessed the exception datastore without
holding ast_channel_lock, which could corrupt the per-channel datastore list
under concurrency and lead to crashes during teardown (e.g. double free in
ast_datastore_free).
Resolves: #1831
The PR #1522 introduced the header read loop for audiosocket packets
which does not handle partial header reads correctly. This commit
adds the missing buffer offsets.
manager.c: Fix CLI event display
- `manager show events`: fix event names being truncated at 20 characters, widen column to 28 to accommodate the longest registered event name
- `manager show events`: skip duplicate entries caused by multiple modules registering the same event name, list is already sorted so adjacent name comparison is sufficient
Redirects initiated by 302 response codes weren't handled correctly
when setting the hangup cause code and tech cause code on the responding
channel. They're now set to 23 (REDIRECTED_TO_NEW_DESTINATION) and
302 (Moved permanently). Other non-2XX response codes also had issues.
A new API ast_channel_dialed_causes_iterator() was added to retrieve
the hangup cause codes for a channel.
chan_pjsip_session_end() in chan_pjsip has been refactored to set the
correct cause codes on a channel based on the cause codes added by
chan_pjsip_incoming_response_update_cause(). Copious amounts of
debugging and comments were also added.
Resolves: #1819
Currently, endpoints created via the PJSIP Config Wizard do not update
their ACL rules if the underlying Named ACL (in acl.conf) changes.
This occurs because the wizard relies on file timestamp and content
caching of pjsip_wizard.conf, which remains unchanged during an external
ACL update. As a result, endpoints retain stale ACL rules even after
a reload.
This patch updates res_pjsip_config_wizard to subscribe to the
ast_named_acl_change_type Stasis event. A local generation counter is
incremented whenever an ACL change event is received.
During a reload, the wizard compares the current local generation against
the generation stored in the wizard object. If a change is detected:
1. The file cache optimization (CONFIG_FLAG_FILEUNCHANGED) is bypassed.
2. Wizard objects utilizing 'acl' or 'contact_acl' are forced to update,
ensuring they pick up the new IP rules.
Signed-off-by: Michal Hajek michal.hajek@daktela.comFixes: #1641
If a channel in Stasis/ARI is hung up by the channel driver, the RTPAUDIOQOS
variables are set before the channel leaves Stasis and are therefore
available to the ARI app via ChannelVarset events. If the channel is hung up
by ARI however, the channel leaves Stasis before the RTPAUDIOQOS variables
are set so the app may not get the ChannelVarset events.
We now set the RTPAUDIOQOS variables when ast_softhangup() is called as well
as when the channel driver hangs up a channel. Since ARI hangups call
ast_softhangup(), the variables will be set before the channel leaves Stasis
and the app should get the ChannelVarset events.
ast_rtp_instance_set_stats_vars(), which actually sets the variables, now
checks to see if the variables are already set before attempting to set them.
This prevents double messages from being generated.
Resolves: #1816
Description:
When Asterisk is running without a timing module, attempting to process DTMF
triggers a segmentation fault. This occurs because the system
attempts to access a null timing file descriptor when setting up the
DTMF emulation timer.
This fix ensures that the system checks for a valid timing source before
attempting to start the DTMF emulation timer. If no timing module is
present, it logs a warning and skips the emulation instead of crashing
the process.
Changes:
- Modified main/channel.c to add a safety check within the __ast_read function.
- Implemented a graceful return path when no timing source is available
- Added a LOG_WARNING to inform the administrator that DTMF emulation
was skipped due to missing timing modules.
Testing:
- Disabled all timing_ modules in modules.conf and confirmed with
'timing test'.
- Reproduced the crash by modifying the dialplan with:
exten => 707,1,NoOp(Starting DTMF - No Timing Mode)
same => n,Answer()
same => n,Background(demo-congrats)
same => n,WaitExten(10)
same => n,Hangup()
And calling 707 followed by 1
- Verified that with the fix applied, the system logs "No timing module
loaded; skipping DTMF timer" and continues dialplan
execution without crashing.
- Confirmed stability during concurrent media sessions and DTMF input.
Fixes: #566
If a pjsip transport (A) fails to load, its temporary state gets left behind
causing the next transport to load (B) to pick up some of its parameters,
including its name. This can cause B to have the correct name (B) in its
transport object but the wrong name (A) in its internal state object. When a
transport state is searched for later on, transport state B is returned but a
retrieval of the actual transport object will fail because B's transport
state id is actually "A" and transport "A" doesn't exist because it failed
to load.
remove_temporary_state() is now being called in all error paths in
config_transport.c functions that call find_or_create_temporary_state().
A bit of extra debugging was also added to res_pjsip_nat.c.
Resolves: #1814
According to RFC 3428 (Section 5), a Contact header is not required in a
MESSAGE request unless the sender wants to establish a session. This
patch ensures that the Contact header is removed from out-of-dialog
MESSAGE requests within res_pjsip_messaging.c.
Fixes: #1356
Add uri prefix based acl support to the built in http server.
This allows an acl to be added per uri prefix (ie '/metrics'
or '/ws') to restrict access.
Add user based acl support for ARI. This adds new acl options
to the user section of ari.conf to restrict access on a per
user basis.
resolves: #1799
UserNote: A new section, type=restriction has been added to http.conf
to allow an uri prefix based acl to be configured. See
http.conf.sample for examples and more information.
The user section of ari.conf can now contain an acl configuration
to restrict users access. See ari.conf.sample for examples and more
information
