The reference identifier (what the client provides - in this case a
hostname) must start with a domain label, not a `.`.
The current implementation will match `.seanbright.com` against
`*.seanbright.com` which is incorrect.
If the channel is locked when calling ast_set_variables and any of the
variables contained dialplan functions, there's a possiblilty of a deadlock.
To prevent this, either the explicit locks were removed or the call to
ast_set_variables moved out of the lock scope. A warning to not hold
channel locks is also added to the documentation for ast_set_variables.
Resolves: #1936
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>
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
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
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.
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
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
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.
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.
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
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
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.
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 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
When TURN is configured in rtp.conf, pjproject re-injects TURN packets
via 127.0.0.1 (the loopback address). The DTLS packet handler checks the
source address against the ICE active candidate list before the loopback
address substitution runs, causing the packet to be silently dropped as
the source 127.0.0.1 is not in the candidate list.
Fix by performing the loopback address substitution before the ICE
candidate source check in the DTLS path, mirroring the logic already
present in the non-DTLS RTP path.
Fixes: #1795
UserNote: WebRTC calls using TURN configured in rtp.conf (turnaddr,
turnusername, turnpassword) will now correctly complete DTLS/SRTP
negotiation. Previously all DTLS packets were silently dropped due to
the loopback re-injection address not being in the ICE active candidate
list.
The AudioSocket encapsulation for externalMedia requires a UUID to be
provided in the `data` parameter of the ARI call. If not provided, we
should return a 400 Bad Request instead of a 500 Internal Server
Error.
Pointed out by AVT in the community forum[1].
1: https://community.asterisk.org/t/externalmedia-audiosocket-on-asterisk-22/112149
There is a LOT of work in this commit but the TL;DR is that it takes
CEL processing from using 38% of the CPU instructions used by a call,
which is more than that used by the call processing itself, down to less
than 10% of the instructions.
So here's the deal... cdr_custom, cdr_sqlite3_custom, cel_custom
and cel_sqlite3_custom all shared one ugly trait...they all used
ast_str_substitute_variables() or pbx_substitute_variables_helper()
to resolve the dialplan functions used in their config files. Not only
are they both extraordinarily expensive, they both require a dummy
channel to be allocated and destroyed for each record written. For CDRs,
that's not too bad because we only write one CDR per call. For CELs however,
it's a disaster.
As far as source code goes, the modules basically all did the same thing.
Unfortunately, they did it badly. The config files simply contained long
opaque strings which were intepreted by ast_str_substitute_variables() or
pbx_substitute_variables_helper(), the very functions that ate all the
instructions. This meant introducing a new "advanced" config format much
like the advanced manager event filtering added to manager.conf in 2024.
Fortunately however, if the legacy config was recognizable, we were able to
parse it as an advanced config and gain the benefit. If not, then it
goes the legacy, and very expensive, route.
Given the commonality among the modules, instead of making the same
improvements to 4 modules then trying to maintain them over time, a single
module "res_cdrel_custom" was created that contains all of the common code.
A few bonuses became possible in the process...
* The cdr_custom and cel_custom modules now support JSON formatted output.
* The cdr_sqlite_custom and cel_sqlite3_custom modules no longer have
to share an Sqlite3 database.
Summary of changes:
A new module "res/res_cdrel_custom.c" has been created and the existing
cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules
are now just stubs that call the code in res_cdrel_custom.
res_cdrel_custom contains:
* A common configuration facility.
* Getters for both CDR and CEL fields that share the same abstraction.
* Formatters for all data types found in the ast_cdr and ast_event
structures that share the same abstraction.
* Common writers for the text file and database backends that, you guessed it,
share the same abstraction.
The result is that while there is certainly a net increase in the number of
lines in the code base, most of it is in the configuration handling at
load-time. The run-time instruction path length is now significanty shorter.
```
Scenario Instructions Latency
=====================================================
CEL pre changes 38.49% 37.51%
CEL Advanced 9.68% 6.06%
CEL Legacy (auto-conv to adv) 9.95% 6.13%
CEL Sqlite3 pre changes 39.41% 39.90%
CEL Sqlite3 Advanced 25.68% 24.24%
CEL Sqlite3 Legacy (auto conv) 25.88% 24.53%
CDR pre changes 4.79% 2.95%
CDR Advanced 0.79% 0.47%
CDR Legacy (auto conv to adv) 0.86% 0.51%
CDR Sqlite3 pre changes 4.47% 2.89%
CEL Sqlite3 Advanced 2.16% 1.29%
CEL Sqlite3 Legacy (auto conv) 2.19% 1.30%
```
Notes:
* We only write one CDR per call but every little bit helps.
* Sqlite3 still takes a fair amount of resources but the new config
makes a decent improvement.
* Legacy configs that we can't auto convert will still take the
"pre changes" path.
If you're interested in more implementation details, see the comments
at the top of the res_cdrel_custom.c file.
One minor fix to CEL is also included...Although TenantID was added to the
ast_event structure, it was always rendered as an empty string. It's now
properly rendered.
UserNote: Significant performance improvements have been made to the
cdr_custom, cdr_sqlite3_custom, cel_custom and cel_sqlite3_custom modules.
See the new sample config files for those modules to see how to benefit
from them.
Currently, WebSockets both accept and send media without the option to
disable one or the other. This commit adds the ability to set the media
direction for a WebSocket, making it unidirectional or bidirectional
(the default). Direction is done from the point of view of the
application, NOT Asterisk. The allowed values are 'both', 'in', and
'out'. If media direction is 'both' (the default), Asterisk accepts and
sends media to the application. If it is 'in', Asterisk will drop any
media received from the application. If it is 'out', Asterisk will not
write any media frames to the application.
UserNote: WebSocket now supports media direction, allowing for
unidirectional media. This is done from the perspective of the
application and can be set via channel origination, external media, or
commands sent from the application. Check out
https://docs.asterisk.org/Configuration/Channel-Drivers/WebSocket/ for
more.
`ast_cli_unregister_multiple()` expects internal data members to be heap
allocated which happens during a successful call to
`ast_cli_register_multiple()`. CLI handlers defined traditionally - those whose
handler responds to the CLI_INIT message - take care of this automatically. But
when we statically provide a `command` or `usage` member, we _must_ initialize
them with `ast_cli_register_multiple()` before attempting to destroy them.
Resolves: #1651
According to RFC 8866 (Section 5.2), the Session Name (s=) field and
the username part of origin (o=) are both mandatory and cannot be
empty. If a session has no name, or no username part of origin, the
RFC recommends using a single dash (-) as a placeholder.
This fix ensures that if the session name or the username part of
origin length is zero , it defaults to -.
Fixes: #1524
This hook may not be necessary when we do a retransmit, but when there are two
INVITEs, one *initial* and one with auth digest, the second INVITE contains wrong (unmodified) media address
due to the commented line below.
The NAT hook needs to run due to filters potentially reverting previously modified packets.
Fixes: #449
GCC 15.2.1 pays attention to the discarding of the const
qualifier when strchr, strrchr, memchr, or memrchr are now
used. This change fixes numerous errors with this throughout
the tree. The fixes can be broken down into the following:
1. The return value should be considered const.
2. The value passed to strchr or strrchr can be cast as it is
expected and allowed to be modified.
3. The pointer passed to strchr or strrchr is not meant to be
modified and so the contents must be duplicated.
4. It was declared const and never should have been.
Adds a new PJSIP_INHERITABLE_HEADER dialplan function to add
inheritable headers from the inbound channel to an outbound
bridged channel. This works similarly to the existing
PJSIP_HEADER function, but will set the header on the bridged
outbound channel's INVITE upon Dial.
Inheritable headers can be updated or removed from the inbound
channel as well as from a pre-dial handler
Resolves: #1670
UserNote: A new PJSIP_HEADER option has been added that allows
inheriting pjsip headers from the inbound to the outbound bridged
channel.
Example- same => n,Set(PJSIP_INHERITABLE_HEADER(add,X-custom-1)=alpha)
will add X-custom-1: alpha to the outbound pjsip channel INVITE
upon Dial.
When an outbound INVITE transaction times out (408) or receives a 503 error,
check_request_status() attempts to failover to the next available address by
restarting the INVITE session. However, the function did not check if the
inv_session was already cancelled before attempting the failover.
This caused unexpected behavior when a caller hung up during a ring group
scenario: after CANCEL was sent but the remote endpoint failed to respond
with 487 (e.g., due to network disconnection), the transaction timeout
would trigger a NEW outbound INVITE to the next address, even though the
session was already terminated.
This violates RFC 3261 Section 9.1 which states that if no final response
is received after CANCEL within 64*T1 seconds, the client should consider
the transaction cancelled and destroy it, not retry to another address.
The fix adds a check for both PJSIP_INV_STATE_DISCONNECTED and inv->cancelling
at the beginning of check_request_status(). This ensures that:
- Failover is blocked when the user explicitly cancelled the call (CANCEL sent)
- Failover is still allowed for legitimate timeout/503 scenarios where no
CANCEL was initiated (e.g., SRV failover when first server is unreachable)
Resolves: #1716
allocate_subscription() increments the ao2 reference count of the subscription tree,
but the reference was not consistently released during subscription destruction,
resulting in leaked sip_subscription_tree objects.
This patch makes destroy_subscription() responsible for releasing sub->tree,
removes ad-hoc cleanup in error paths,
and guards tree cleanup to ensure refcount symmetry and correct ownership.
Fixes: #1703
Reduce cache lock time for AMI and CLI sorcery memory cache populate
commands by adding a new populate_lock to the sorcery_memory_cache
struct which is locked separately from the existing cache lock so that
the cache lock can be maintained for a reduced time, locking only when
the cache objects are removed and re-populated.
Resolves: #1700
UserNote: The AMI command sorcery memory cache populate will now
return an error if there is an internal error performing the populate.
The CLI command will display an error in this case as well.
We now add the websocket's file descriptor to the channel's fd array and let
it poll for data availability instead if having a dedicated thread that
does the polling. This eliminates the thread and allows removal of most
explicit locking since the core channel code will lock the channel to prevent
simultaneous calls to webchan_read, webchan_hangup, etc.
While we were here, the hangup code was refactored to use ast_hangup_with_cause
instead of directly queueing an AST_CONTROL_HANGUP frame. This allows us
to set hangup causes and generate snapshots.
For a bit of extra debugging, a table of websocket close codes was added
to http_websocket.h with an accompanying "to string" function added to
res_http_websocket.c
Resolves: #1683
Always take an explicit reference on the format used for bridge playback
and recording channels, regardless of where it was sourced, and release
it after prepare_bridge_media_channel. This aligns the code paths and
avoids mixing borrowed and owned references while preserving behavior.
Fixes: #1648
* 3d positions were being rendered without an enclosing `<gml:pos>`
element resulting in invalid XML.
* There was no way to set the `id` attribute on the enclosing `tuple`, `device`
and `person` elements.
* There was no way to set the value of the `deviceID` element.
* Parsing of degree and radian UOMs was broken resulting in them appearing
outside an XML element.
* The UOM schemas for degrees and radians were reversed.
* The Ellipsoid shape was missing and the Ellipse shape was defined multiple
times.
* The `crs` location_info parameter, although documented, didn't work.
* The `pos3d` location_info parameter appears in some documentation but
wasn't being parsed correctly.
* The retransmission-allowed and retention-expiry sub-elements of usage-rules
were using the `gp` namespace instead of the `gbp` namespace.
In addition to fixing the above, several other code refactorings were
performed and the unit test enhanced to include a round trip
XML -> eprofile -> XML validation.
Resolves: #1667
UserNote: Geolocation: Two new optional profile parameters have been added.
* `pidf_element_id` which sets the value of the `id` attribute on the top-level
PIDF-LO `device`, `person` or `tuple` elements.
* `device_id` which sets the content of the `<deviceID>` element.
Both parameters can include channel variables.
UpgradeNote: Geolocation: In order to correct bugs in both code and
documentation, the following changes to the parameters for GML geolocation
locations are now in effect:
* The documented but unimplemented `crs` (coordinate reference system) element
has been added to the location_info parameter that indicates whether the `2d`
or `3d` reference system is to be used. If the crs isn't valid for the shape
specified, an error will be generated. The default depends on the shape
specified.
* The Circle, Ellipse and ArcBand shapes MUST use a `2d` crs. If crs isn't
specified, it will default to `2d` for these shapes.
The Sphere, Ellipsoid and Prism shapes MUST use a `3d` crs. If crs isn't
specified, it will default to `3d` for these shapes.
The Point and Polygon shapes may use either crs. The default crs is `2d`
however so if `3d` positions are used, the crs must be explicitly set to `3d`.
* The `geoloc show gml_shape_defs` CLI command has been updated to show which
coordinate reference systems are valid for each shape.
* The `pos3d` element has been removed in favor of allowing the `pos` element
to include altitude if the crs is `3d`. The number of values in the `pos`
element MUST be 2 if the crs is `2d` and 3 if the crs is `3d`. An error
will be generated for any other combination.
* The angle unit-of-measure for shapes that use angles should now be included
in the respective parameter. The default is `degrees`. There were some
inconsistent references to `orientation_uom` in some documentation but that
parameter never worked and is now removed. See examples below.
Examples...
```
location_info = shape="Sphere", pos="39.0 -105.0 1620", radius="20"
location_info = shape="Point", crs="3d", pos="39.0 -105.0 1620"
location_info = shape="Point", pos="39.0 -105.0"
location_info = shape=Ellipsoid, pos="39.0 -105.0 1620", semiMajorAxis="20"
semiMinorAxis="10", verticalAxis="0", orientation="25 degrees"
pidf_element_id = ${CHANNEL(name)}-${EXTEN}
device_id = mac:001122334455
Set(GEOLOC_PROFILE(pidf_element_id)=${CHANNEL(name)}/${EXTEN})
```
Delay acquisition of the ast_sip_endpoint reference in mwi_get_notify_data()
to avoid an ao2 ref leak on early-return error paths.
Move ast_sip_subscription_get_endpoint() to just before first use so all
acquired references are properly cleaned up.
Fixes: #1675
This commit integrates the redirect module into res_pjsip_messaging
to enable following 3xx redirect responses for outgoing SIP MESSAGEs.
When follow_redirect_methods contains 'message' on an endpoint, Asterisk
will now follow 3xx redirect responses for MESSAGEs, similar to how
it behaves for INVITE responses.
Resolves: #1576
UserNote: A new pjsip endpoint option follow_redirect_methods was added.
This option is a comma-delimited, case-insensitive list of SIP methods
for which SIP 3XX redirect responses are followed. An alembic upgrade
script has been added for adding this new option to the Asterisk
database.
This commit introduces a new redirect handling module that provides
infrastructure for following SIP 3xx redirect responses. The redirect
functionality respects the endpoint's redirect_method setting and only
follows redirects when set to 'uri_pjsip'. This infrastructure can be
used by any PJSIP module that needs to handle 3xx redirect responses.
Sean Bright
George Joseph
Jaco Kroon
Maximilian Fridrich
Mike Bradeen
Peter Krall
Sebastian Denz
mattia
phoneben
Daniel Donoghue
Sven Kube
nappsoft
Sven Kube
Michal Hajek
Alexis Hadjisotiriou
Robert Wilson
Ben Ford
Prashant Srivastav
Arcadiy Ivanov
Naveen Albert
Joshua C. Colp
hishamway
Alexei Gradinari