asterisk

Commit Graph

Author	SHA1	Message	Date
Gian Diego Javes	6d383d321d	taskpool: count the in-flight task when selecting an executor The taskpool selectors measure load with ast_taskprocessor_size(), which only reports queued tasks. A taskprocessor that is executing a long running task has an empty queue, so it looks identical to an idle one and the least full selector hands it more work while other executors sit free. Add ast_taskprocessor_is_executing() and include the in-flight task in the load the selectors compare. This is not a new semantic. taskprocessor_push() already counts the in-flight task as outstanding work when it notifies the listener: /* The currently executing task counts as still in queue */ was_empty = tps->executing ? 0 : previous_size == 0; The taskpool selectors are the one place that still measures only the queue. This change applies the same rule there. The accessor reads tps->executing without the object lock, the way the CLI already does in main/taskprocessor.c, because taking the lock on every push cost roughly six times the throughput: taskpool push efficiency went from 324438 to 52966 tasks per second and the serializer variant from 217467 to 38313. Without the lock both are back to baseline. A stale read only means one push is routed as if the taskprocessor had just changed state, which the selector already tolerates. Also add a unit test that holds one executor of a four executor pool inside a long task and checks that a task pushed to a different serializer runs on one of the three free executors. It fails on every run without this change. Fixes: #2074 UserNote: Taskpool executors that are running a long task are no longer considered idle when work is distributed, so tasks are routed to free executors instead of queueing behind a busy one.	2 months ago
aabolfazl	fccb7cb367	app_voicemail: Fix crash when VM_INFO reads an unset email address. The email member of struct ast_vm_user is a pointer rather than a fixed array, and populate_defaults() leaves it NULL when a mailbox has no email address configured. Every other attribute VM_INFO reads is a fixed array, so only the email attribute is affected. VM_INFO passed vmu->email straight to ast_copy_string(), which dereferences its source unconditionally. Reading the email attribute of a mailbox that has no email address therefore crashed Asterisk from the dialplan. Guard the copy with S_OR() so an unset email address yields an empty string, matching how the language attribute already handles its fallback. The make_email_file() call sites were already guarded and are left alone. Add a regression test for the unset case, and restore the voicemail configuration when the VM_INFO test finishes. That test was the only one in app_voicemail that did not do so, which left its test mailbox in the users list and made the test fail if it ran a second time. Fixes: #2063	2 months ago
Sean Bright	9579b24a0a	res_agi.c: Prevent out-of-bounds array access when parsing arguments The `parse_args(...)` function writes to the `argv` array in 2 locations but was only bounds checking in one of them. Moved the bounds check so that it is encountered on each iteration through the parsing loop. Resolves: #2069	2 months ago
Mike Bradeen	f0c1c1a7b7	res_pjsip_sdp_rtp: Fix intermittent call drop on reINVITE sdp change A reINVITE that changed the offered codec set (e.g. ulaw+alaw -> alaw only) could lead to call teardown when a bridge write still using the old format occurred during the re-negotiation. This change avoids the termination by making the following changes. First, Asterisk now holds the lock across both the tx payload update and the channel format update so they commit atomically. Second, when processing the sdp on the new incoming offer, merge the new offer with the old until the negotiation is complete. Also adds unit tests for the sdp merge Fixes: #2059	2 months ago
Sven Kube	926a1068f2	chan_websocket: Fix NULL requestor dereference in webchan_request. Two log statements in webchan_request() passed `requestor` straight to ast_channel_name(), which dereferences the channel with no NULL check. `requestor` is NULL whenever no originator channel is supplied: ARI POST /channels/externalMedia always passes NULL, and POST /channels/create passes NULL when `originator` is omitted.	2 months ago
Naveen Albert	f6b02c640c	backtrace.c: Include stdbool.h stdbool.h is needed now that we use the bool type directly instead of bfd_boolean. See also 1d95b744c06974f0a00c143c6e0fc979af930908. Resolves: #2061	2 months ago
aabolfazl	c46c9945a8	res_musiconhold: Fix mohclass reference leak on answeredonly early return. local_ast_moh_start() returns -1 from the answeredonly check without releasing the mohclass reference it holds, unlike every other exit path in the function. Nothing else ever releases that reference, so moh_class_destructor() never runs for the object. With realtime music on hold and cachertclasses disabled, each suppressed request leaks the class object, its monitor thread, the external application process and two file descriptors for the lifetime of Asterisk. For static classes the stale references prevent the class from ever being destroyed after it is replaced by a reload. Release the reference before returning, matching the other exit paths. Fixes: #2051	2 months ago
Naveen Albert	5b59829015	backtrace.c: Avoid removed bfd_boolean type. bfd_boolean has been deprecated for some time and has now been removed in gdb, so use bool directly instead. See: `1d95b744c0` Resolves: #2054	2 months ago
Sean Bright	73064ff3bf	app_voicemail.c: Fix may-be-uninitialized error A pointer to `new` is passed to `inboxcount(...)` which increments the pointed-to value.	2 months ago
Mike Bradeen	2720b1025e	pjproject: disable building pjsua when a sanitizer is used Disable building the pjsua test application when any of the SANITIZER flags are set. Fixes: #2056	2 months ago
George Joseph	b531f71b12	SECURITY.md: Add warning about reporting multiple issues in one advisory.	2 months ago
Mehrdad Seifzadeh	8cd58fd39d	chan_pjsip: Store transport info for outgoing channels Outgoing PJSIP channels did not have transport information stored in their session datastore when CHANNEL(pjsip,local_addr) or CHANNEL(pjsip,remote_addr) was read. As a result, the address fields were empty on B-leg channels even though other dialog fields such as call-id and URIs were available. Store the selected local transport address and destination address for outgoing UAC session requests once the request is transmitted and the transport information is known. Fixes: #871	2 months ago
aabolfazl	c05ac3f232	func_strings: Fix misspelling of "occurred" in FIELDNUM documentation. The FIELDNUM description misspelled "occurred" as "occured". This text is user-visible: it is rendered on docs.asterisk.org and printed by "core show function FIELDNUM".	2 months ago
Jeremy Lainé	7c60726df1	tcptls: Don't inhibit escalations for outbound client connections. handle_tcptls_connection() marks the current thread as inhibiting privilege escalations and as an external user interface so that dialplan functions considered 'dangerous' (STAT, SHELL, ...) cannot be executed on behalf of an external protocol. This is correct for inbound (server) connections, each of which runs on its own dedicated worker thread. However, the outbound (client) path calls handle_tcptls_connection() synchronously on the caller's own thread. When that caller is a channel/PBX thread, the thread-local flags are set and never cleared, permanently tainting the dialplan thread. Examples: - Calling Dial() for an outbound WebSocket and resuming dialplan execution with the "g" flag. - Running ExternalIVR() then continuing dialplan executing. Outbound connections are initiated by Asterisk itself and are not external user interfaces, so the flags should not be set for them. Gate the flag-setting on tcptls_session->client so it applies only to inbound connections. Resolves: #2038	2 months ago
Mike Bradeen	734fb592bd	res_rtp_asterisk: Avoid two lock inversion deadlocks with pj project Fixes two related instance, group lock inversion deadlocks. When writing the RTP stream via rtp_sendto, the RTP instance was only unlocked before being passed to pj project when the instance was un-bundled (ie video was not bundled onto the audio stream's ICE transport.) The first change unlocks the instance whether bundled or unbundled to avoid the deadlock, then re-checks the bundled status upon return to know wether or not the transport should be used. For the second change, when an incoming RTCP NACK record was recieved, the parent (transport) and child (instance) locks were both held before the call to rtp_sendto, which is only able to release the child lock. Now in the un-bundled case the parent and child will be unlocked and re-locked in the correct order via a new helper function before and after the call to send_to. Fixes: #1946	2 months ago
Stanislav Abramenkov	05dbd160ea	jansson: Upgrade version to jansson 2.15.1 UpgradeNote: jansson has been upgraded to 2.15.1. For more information visit jansson Github page: https://github.com/akheron/jansson/releases/tag/v2.15.1 Resolves: #2033	2 months ago
Jeremy Lainé	b31d3b1d57	file.c: Ensure opening a stream opens at most one file The functions used to stream files eventually end up calling into `filehelper(.., ACTION_OPEN)` which takes care of iterating over supported extensions and opening the first existing file. To do this, `filehelper` uses two nested loops: - An outer loop over the supported file formats. - An inner loop over the possible extensions for that format. We need to break out of both loops as soon as a file was successfully opened. Otherwise if a file exists in multiple formats (e.g `foo.wav` and `foo.alaw`) both files will be opened successively. Resolves: #1625	2 months ago
Joshua C. Colp	ad429910c6	pjsip: Eliminate some unique taskprocessors. When placing outbound calls each call would get its own taskprocessor for things related to the call. In practice this is overkill as few things actually occur. Inbound calls on the other hand already use one of the fixed number of distributor taskprocessors. This also occurred for OPTIONS requests with each AOR having its own taskprocessor. This change moves both to using a distributor taskprocessor instead.	2 months ago
Roel van Meer	e22687d0bd	bridging: Add support for TOUCH_MIXMONITOR_OPTIONS This channel variable can be used to override the default automixmon options (which are 'b' or 'bB(<n>)'). If both TOUCH_MIXMONITOR_OPTIONS and TOUCH_MIXMONITOR_BEEP are set, TOUCH_MIXMONITOR_BEEP is ignored with a warning. It is the responsibility of the user to configure valid options in TOUCH_MIXMONITOR_OPTIONS. Fixes: #1981 UserNote: The TOUCH_MIXMONITOR_OPTIONS channel variable can now be used to configure the options used by the mixmonitor application when it is started via automixmon. If both TOUCH_MIXMONITOR_OPTIONS and TOUCH_MIXMONITOR_BEEP are set, TOUCH_MIXMONITOR_BEEP is ignored with a warning.	2 months ago
George Joseph	1480af1227	res_pjsip_refer: Fix issues with Refer-Sub:false and ARI. When an attended transfer is controlled by ARI using PJSIP_TRANSFER_HANDLING()=ari-only, and a REFER request is received with a Refer-Sub: false header to suppress the automatic progress subscription, a segfault occurs in res_pjsip_refer:refer_incoming_ari_request(). However, even if the segfault were prevented, there's another issue... Suppressing the subscription and NOTIFYs to the referer also prevents the ARI app from getting any progress events for the transfer which it's controlling. So... * The segfault has been fixed. * Suppressing the subscription no longer suppress the ARI events from being sent to the controlling ARI app. * A good amount of tracing has also been added. Resolves: #2021	2 months ago
Mike Bradeen	ea9b48659d	build: re-add pjsua test application pjsua and it's python bindings were removed as part of PR 1854. Testing for RTT requires the pjsua application be re-added to the third pary build process. This applies to the pjsua application ONLY and not the associated python bindings. Resolves: #2024	2 months ago
Naveen Albert	036f09c073	res_pjsip: Document 'none' option for 'dtmf_mode'. The 'none' value can be used to disable DSP processing for DTMF on PJSIP channels and is sometimes necessary for this reason, and already exists in the code, but is not documented. Add it to the documentation enum. Resolves: #2026	2 months ago
Mike Bradeen	fb4bfb1156	Audiohooks: Allow whisper audiohooks to work without an underlying stream Audiohook whispers work by mixing audio with the current stream of frames being written to the hooked channel. Prior to this change if there were no audio frames being written to the channel outside of the audiohook, then any whisper audio has nothing to mix with and so is not heard on the hooked channel. This change removes that requirement by creating a framehook driven timer on the channel that creates silent frames the whispered frames can be mixed into. The framehook is automatically added with the first attached whisper audiohook and is removed when the last one is detached. If there are no attached streams being written to the channel, the timer is used to generate a default one. If a stream is attached; the timer is ignored until the stream stops. This stop and start is automatically triggered by the presence of a written stream and does not require any external action. Resolves: #1966 UserNote: There is no longer a requirement to play silence or otherwise have a stream of audio being sent to a hooked channel for whispered frames to be heard.	2 months ago
Ben Ford	99fe892b66	Logger: Don't discard WARNING or ERROR messages. When the logger reaches it's threshold for messages, don't discard the WARNING or ERROR messages up to a certain amount. If THAT threshold reaches its maximum, start discarding the oldest non-WARNING/ERROR messages until we only have WARNING and ERROR messages left. Everything will be discarded after that until we have room again for more. Also added a test that limits the queue sizes and logs many messages to see the queues hitting their maximums and discarding the appropriate messages. Fixes: #1882	2 months ago
Aqeel Abbas	95b94d0ef1	func_strings: Fix syntax error in STRBETWEEN documentation example The STRBETWEEN function documentation example was missing a closing brace in the SendDTMF call. Corrected the syntax to ensure the dialplan example is functional.	2 months ago
Mehrdad Seifzadeh	07884c4144	res_pjsip_registrar: Resolve AOR for identified endpoints When an inbound REGISTER is matched to an endpoint using an identification method such as IP, header, or request URI, the endpoint can be identified and authenticated successfully while registrar AOR resolution still fails. find_registrar_aor() only resolved AOR names for username and auth_username identify methods. A header-only endpoint therefore left aor_name unset and failed registration with "AOR '' not found" even when the REGISTER To URI matched one of the endpoint's configured AORs. Resolve the REGISTER AOR from the To URI for endpoint identification methods that do not directly provide an AOR name, while still constraining the match to the already-identified endpoint's configured AOR list. Fixes: #1398	2 months ago
Thomas Guebels	ae85ad744a	res_http_websocket: Unref session when it fails to establish. Two error paths in res_http_websocket could leak an ast_websocket session by failing to drop a reference after the session had been allocated. The first occurs when session ID generation fails. While this is largely theoretical, as it would require an out-of-memory condition, the session reference should still be released correctly. The second occurs when the ast_websocket_pre_callback session_attempted callback returns an error. This path is reachable through ari_websockets, which implements this callback and can legitimately fail under certain conditions. Fixes: #2020	3 months ago
George Joseph	546749516b	Bundled pjproject: Make it easier to override options in config_site.h. Bundled pjproject uses ./third-party/pjproject/patches/config_site.h to set many pjproject build options like PJ_IOQUEUE_MAX_HANDLES, PJSIP_MAX_URL_SIZE, etc. Editing that file however, causes the git tree to become dirty which is inconvenient. So... * Updated the Asterisk configure scripts to pass the PJPROJECT_CFLAGS variable down to the pjproject configure scripts. See the UserNote below for details. * Updated ./third-party/pjproject/patches/config_site.h to allow the following options to be overridden: PJ_MAX_HOSTNAME, PJSIP_MAX_URL_SIZE, PJ_IOQUEUE_MAX_HANDLES. Other options in config_site.h are not overridable because they can have a bad effect on the overall operation of pjproject. This may be revisited in the future. Options not already set in config_site.h can still be set. * Fixed an issue where if the Linux `epoll` facility is used (which it is by default) the default PJ_IOQUEUE_MAX_HANDLES and PJSIP_MAX_TRANSPORTS were being left at 1024 instead of being icnreased to 5000. * The `pjproject show buildopts` CLI command previously only showed options from the top level `PJ` pjproject layer but now also shows many from the `PJSIP` layer. Many of these, such as PJSIP_MAX_TRANSPORTS, can be set using PJPROJECT_CFLAGS provided they're not already unconditionally set in config_site.h. * The `pjsip dump endpt` CLI command previously required that the pjproject log level be already set to at least 3 or no output would be produced. The command now does that automatically then sets it back to whatever it was. This isn't strictly related to this PR but was just nagging me. UserNote: Bundled pjproject: It's now possible to override some of the pjproject build options contained in ./third-party/pjproject/patches/config_site.h by adding PJPROJECT_CFLAGS to your Asterisk ./configure command line. For example: `./configure ... PJPROJECT_CFLAGS='-DPJ_OPT1=8192 -DPJ_OPT2=512'` Any option in config_site.h that's wrapped in a `#ifndef` block can be overridden and many of the `PJSIP` options displayed by `pjproject show buildopts` can be set. WARNING: Adjusting these options without understanding their effect can cripple your Asterisk instances. You shouldn't adjust them unless you need to solve a specific issue. Resolves: #2011	3 months ago
Joshua C. Colp	1fc956a48b	test_performance: Add performance experimentation test module. This is a module which is only built when TEST_FRAMEWORK is enabled and provides CLI commands for testing performance of certain things on the system they are invoked on. The first CLI commands added cover the most common container usage: storage of objects with a lookup based on a string key. These commands take various arguments and allow you to see how they perform. There is also an "all" command named "performance test container_key_lookup_all" that will execute all of these container tests and pass through the given arguments, which makes it easy to run all of the tests for given usage. To facilitate a vector bsearch test a new AST_VECTOR_BSEARCH macro has been added that allows more efficient searching of sorted vectors.	3 months ago
Joshua C. Colp	2158dff4b5	manager: Move away from shared linked list for events. This change moves manager away from a shared linked list for events to a per-session vector of pending events. This allows early filtering of events which eliminates the need to wake up threads for events that they have no interest in. TCP based connections have also been moved to an alert pipe based wakeup mechanism to reduce locking and simplify usage. UserNote: The "manager show eventq" CLI command has been removed as there is no longer a single event queue to display.	3 months ago
George Joseph	5eded44fee	chan_websocket: Use leftover data if no frames are available when the timer fires. When the 20ms channel timer fires but there are no frames available in the queue, we now check for leftover data in the buffer and if there is any, we create a frame with it and send it to the core. This resolves an issue with the leftover data being delayed if a STOP_MEDIA_BUFFERING command is delayed. Some existing comments were also clarified to account for the new behavior. Resolves: #2006	3 months ago
Mehrdad Seifzadeh	f7c50cff7c	res_pjsip_session: Bound delayed BYE behind UAC INVITE When a confirmed session is being terminated while an outgoing in-dialog INVITE transaction is still outstanding, the BYE is delayed until the outstanding transaction terminates. If that INVITE has already received a provisional response and the final response is malformed and rejected before transaction processing, the transaction can remain outstanding and the delayed BYE can keep the session, media state, RTP instance, and PJPROJECT pools referenced after the channels are gone. When a BYE is delayed behind an outstanding UAC INVITE, set a PJPROJECT transaction timeout on that INVITE so the delayed cleanup path has a bounded wait. If PJPROJECT terminates the dialog as a result of the timeout, discard the delayed BYE instead of sending a duplicate BYE. Fixes: #1965	3 months ago
Sean Bright	a12cc71282	extensions.ael.sample: Restore removed macros Commit `e8f548c1` removed AEL `macro` definition and calls from the sample configuration file, but those do not use the deprecated/removed `Macro` app - they use `Gosub` under the hood.	3 months ago
Alexandre Fournier	a9d33edc2e	format_cap: guard against NULL src in _from_cap helpers ast_format_cap_append_from_cap() and ast_format_cap_replace_from_cap() dereference 'src' (src->preference_order) without checking it for NULL. A dummy channel allocated with ast_dummy_channel_alloc() never sets a native-format capability, so ast_channel_nativeformats() returns NULL on such channels. When CHANNEL(audionativeformat) / CHANNEL(videonativeformat) is evaluated against a dummy channel (e.g. via ARI channelvars during a Stasis VarSet event raised while app_voicemail builds the notification email on a dummy channel), func_channel_read() passes that NULL straight into ast_format_cap_append_from_cap(), causing a NULL dereference at offset 0x28 and a SIGSEGV. Guard both helpers against a NULL source. A NULL source simply means "no formats to copy", so appending/replacing nothing is the correct no-op behaviour. This also protects all other callers. Fixes https://github.com/asterisk/asterisk/issues/1992 AI disclosure: this was generated using Claude Opus 4.8, tested to fix the issue. Not sure if it is the right* way to do it.	3 months ago
ThatTotallyRealMyth	c0fc2ae54c	ast_loggrabber: Install the ast_tsconvert.py script to a secure temp directory. The ast_tsconvert.py script called by ast_loggrabber is now installed in a temporary directory that isn't world readable or writable. Resolves: #GHSA-xgj6-2gc5-5x9c	3 months ago
George Joseph	6f414ec771	chan_unistim.c: Prevent overrun of phone_number field. Add a check to key_dial_page() to ensure that dialed digits won't overrun the phone_number field. Resolves: #GHSA-3g56-cgrh-95p5	3 months ago
Mike Bradeen	42d3552d2a	ooh323c: not checking for IE minimum length When decoding q.931 encoded calling/called number now checking for length being less than minimum required. Resolves: #GHSA-h5hv-jmgj-92q2	3 months ago
George Joseph	762b4c9c39	res_ari: Ensure read-only users are properly authorized via REST Over WebSocket. The REST over WebSocket path now properly prevents non-GET methods from being executed on inbound WebSockets. * The query parameters from the original incoming GET request that caused the upgrade to WebSocket are now passed to all REST requests that come from the client. This ensures that if the client authenticated with a read-only userid using the "api_key" query_string parameter, REST requests coming in over the WebSocket will only be able to execute GETs on resources. The HTTP headers were already passed to the REST requests so if the client had authenticated via an "Authorization" it was properly handled. * New tests have been added to test_ari.c to check that read-only users are properly denied access to resources using non-GET methods. Several memory leaks were also squashed. Resolves: #GHSA-wcvv-g26m-wx5c	3 months ago
George Joseph	f1e4762147	pjsip_message_filter: Use pj_strdup instead of pj_strassign to save local address. The filter_on_tx_message() function was using pj_strassign() to save the pointer of the pjproject transport local address to a local pj_str_t variable. That variable was ultimately used to set the Contact header's uri->host and the SDP connection attribute's address again using pj_strassign. pj_strassign() doesn't copy the actual value of the pj_str_t however, it just copies the pointer so if a connection-oriented transport is disconnected before the 200 OK with the SDP is sent, those pointers will be invalid which can cause use-after-free issues. To prevent this, filter_on_tx_message() now uses pj_strdup with the tdata->pool as the backing store to save the local IP address to the local variable. pj_strassign() can then be used safely later on since the tdata will be available for the life of the transaction. Resolves: #GHSA-g8q2-p36q-94f6	3 months ago
George Joseph	5e59cbad64	ooh323c/ooq931.c: Ensure ooQ931Decode doesn't run out-of-bounds. Several bounds checks have been edded to ooQ931Decode to prevent it from running past the end of the data buffer when parsing information elements. Resolves: #GHSA-746q-794h-cc7f	3 months ago
George Joseph	8f8ccaf70a	ARI: Make ARI applications respect live_dangerously. DeveloperNote: ARI applications can no longer call "dangerous" dialplan functions like DB(), FILE(), SHELL(), CURL(), STAT(), etc. without enabling "live_dangerously" in asterisk.conf. Resolves: #GHSA-vrfp-mg3q-3959	3 months ago
George Joseph	536218a86b	res_rtp_asterisk.c: Address 2 potential T.140 RED buffer overruns. * Add check to red_t140_to_red() to ensure that the new primary payload can't cause the rtp_red->len array items to wrap or cause an overrun of the rtp_red->t140red_data buffer. * Add check to rtp_red_buffer() to ensure that a T.140 frame to be sent can't cause rtp_red->len array items to wrap or cause an overrun of the rtp_red->buf_data buffer. Resolves: #GHSA-vfhr-r9x9-c687 Resolves: #GHSA-j2mm-57pq-jh94	3 months ago
Roberto Paleari	1039bed76f	res/res_pjsip_pubsub.c: Fix buffer over-read in MWI body parser Add constraint checks to prevent unauthenticated users from crashing Asterisk instance by sending a crafted inbound SIP NOTIFY request with "Content-Type: application/simple-message-summary". Resolves: #GHSA-8jw3-ccr9-xrmf	3 months ago
Mike Bradeen	87a1f43e4c	manager: Use remote address in user error logging To avoid a potential null dereference use the remote address in error logging when there is no user or the user acl fails. Resolves: #GHSA-3rhj-hhw7-m6fw	3 months ago
Mike Bradeen	bf0c9a9068	ooh323: Prevent potential buffer overflow in trace logging Replace a call to vsprintf with a call to ast_vasprintf to prevent a possible buffer overflow. Resolves: #GHSA-x348-j6c9-77f3	3 months ago
Pengpeng Hou	59b74276f2	app_sms: Bound protocol 1 SMS unpacking to fixed-size buffers The protocol 1 unpack helpers trusted externally controlled lengths and wrote them directly into fixed-size buffers in sms_t. Clamp the address, header, and body copies to the destination array sizes so malformed messages cannot overwrite adjacent state. Resolves: #GHSA-q9fr-m7g8-6ph5	3 months ago
Milan Kyselica	12072a559f	res_xmpp: Fix stack buffer overflow in namespace prefix handling The snprintf size parameter in xmpp_action_hook() is computed from the attacker-controlled namespace prefix length and is not bounded by the 256-byte stack buffer size. When a remote XMPP peer sends a stanza with a child element whose namespace prefix exceeds 249 characters, snprintf writes past the buffer boundary. Use sizeof(attr) as the snprintf size limit and %.*s precision to extract only the prefix portion of the element name, preserving the original truncation behavior for valid inputs. Resolves: #GHSA-mxgm-8c6f-5p8f	3 months ago
Milan Kyselica	7a1ffcdf38	res_pjsip_pubsub: Add width limit to sscanf in MWI NOTIFY parser The parse_simple_message_summary() function uses sscanf with an unbounded %s format specifier to parse the Message-Account field from incoming SIP NOTIFY bodies into a fixed-size 512-byte stack buffer (PJSIP_MAX_URL_SIZE). A single unauthenticated SIP NOTIFY with a Message-Account value exceeding 512 bytes overflows the buffer, corrupting adjacent stack data and permanently disabling the PJSIP transport layer without crashing the process. Add a width specifier (%511s) to limit the sscanf write to PJSIP_MAX_URL_SIZE - 1 bytes plus the NUL terminator, matching the destination buffer size. Resolves: #GHSA-589g-qgf8-m6mx	3 months ago
Milan Kyselica	04aa482212	res_config_ldap: Escape LDAP filter values per RFC 4515 The LDAP realtime driver constructs search filters by directly concatenating user-supplied values without RFC 4515 escaping. When LDAP is used as a realtime backend for endpoint identification, characters with special meaning in LDAP filters (*, (, ), \) can be injected via the SIP From header username. Add ldap_filter_escape_value() that escapes RFC 4515 special characters to their \HH hex representation, and apply it to non-LIKE query values. The LIKE query path preserves the existing wildcard conversion behavior with a note for maintainers. Resolves: #GHSA-r6c2-hwc2-j4mp	3 months ago
Milan Kyselica	88c65757b5	cel_pgsql, cel_tds: Escape eventtype field to prevent SQL injection The eventtype column handler in cel_pgsql.c inserts record.user_defined_name directly into the SQL query without calling PQescapeStringConn(), while all other string fields in the same function are properly escaped. Similarly, cel_tds.c passes the raw user_defined_name into the SQL INSERT without routing it through anti_injection(), while all other fields are processed through that function. For cel_pgsql.c, escape the eventtype value using PQescapeStringConn(), matching the existing pattern used for all other string fields at lines 308-331 of the same function. For cel_tds.c, route the eventtype value through anti_injection() consistent with how all other fields are handled in the same function. Resolves: #GHSA-ph27-3m5q-mj5m	3 months ago

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