33348 Commits (514f8c3c7ebfe5280fa9fa90f69e76865f667cad)

 

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Author	SHA1	Message	Date
Asterisk Development Team	514f8c3c7e	Update for certified-18.9-cert13	4 months ago
Ben Ford	818dd29a80	manager.c: Restrict ListCategories to the configuration directory. ... When using the ListCategories AMI action, it was possible to traverse upwards through the directories to files outside of the configured configuration directory. This action is now restricted to the configured directory and an error will now be returned if the specified file is outside of this limitation. Resolves: #GHSA-33x6-fj46-6rfh UserNote: The ListCategories AMI action now restricts files to the configured configuration directory.	4 months ago
Asterisk Development Team	d03864911e	Update for certified-18.9-cert12	8 months ago
Jean-Denis Girard	ecd1a14793	app_voicemail: Fix sql insert mismatch caused by cherry-pick ... When commit e8c9cb80 was cherry-picked in from master, the fact that the 20 and 18 branches still had the old "macrocontext" column wasn't taken into account so the number of named parameters didn't match the number of '?' placeholders. They do now. We also now use ast_asprintf to create the full mailbox query SQL statement instead of trying to calculate the proper length ourselves. Resolves: #831 (cherry picked from commit f5845b7d7d)	8 months ago
Sean Bright	42d3038141	alembic: Make 'revises' header comment match reality. ... (cherry picked from commit 13d38eeabe)	8 months ago
Mike Bradeen	03b72fae96	res_pjsip_notify: add dialplan application ... Add dialplan application PJSIPNOTIFY to send either pre-configured NOTIFY messages from pjsip_notify.conf or with headers defined in dialplan. Also adds the ability to send pre-configured NOTIFY commands to a channel via the CLI. Resolves: #799 UserNote: A new dialplan application PJSIPNotify is now available which can send SIP NOTIFY requests from the dialplan. The pjsip send notify CLI command has also been enhanced to allow sending NOTIFY messages to a specific channel. Syntax: pjsip send notify <option> channel <channel> (cherry picked from commit 8e97266810)	8 months ago
George Joseph	d46bc45df9	manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE ... If you run an AMI CoreShowChannelMap on a channel that isn't in a bridge and you're in DEVMODE, you can get a FRACK because the bridge id is empty. We now simply return an empty list for that request. (cherry picked from commit 0778c95e37)	8 months ago
Ben Ford	240597fb26	channel: Add multi-tenant identifier. ... This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID. (cherry picked from commit 9ee00e0d60)	8 months ago
George Joseph	5283b32993	res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback ... The ub_result pointer passed to unbound_resolver_callback by libunbound can be NULL if the query was for something malformed like `.1` or `[.1]`. If it is, we now set a 'ns_r_formerr' result and return instead of crashing with a SEGV. This causes pjproject to simply cancel the transaction with a "No answer record in the DNS response" error. The existing "off nominal" unit test was also updated to check this condition. Although not necessary for this fix, we also made ast_dns_resolver_completed() tolerant of a NULL result. Resolves: GHSA-v428-g3cw-7hv9	8 months ago
Asterisk Development Team	fce1a9b823	Update for certified-18.9-cert11	9 months ago
Mike Bradeen	6b8d8a9823	res_stasis: fix intermittent delays on adding channel to bridge ... Previously, on command execution, the control thread was awoken by sending a SIGURG. It was found that this still resulted in some instances where the thread was not immediately awoken. This change instead sends a null frame to awaken the control thread, which awakens the thread more consistently. Resolves: #801 (cherry picked from commit f974ea2810)	9 months ago
George Joseph	6a2aad1d41	.github: Allow testing an Asterisk PR against a testsuite PR ... (cherry picked from commit a47f92c2f8)	9 months ago
George Joseph	e1962c0911	.github: Add params to Releaser for FPBX issue creation ... (cherry picked from commit 74328a7bfe)	9 months ago
George Joseph	a89e9ca13e	res_pjsip_config_wizard.c: Refactor load process ... The way we have been initializing the config wizard prevented it from registering its objects if res_pjsip happened to load before it. * We now use the object_type_registered sorcery observer to kick things off instead of the wizard_mapped observer. * The load_module function now checks if res_pjsip has been loaded already and if it was it fires the proper observers so the objects load correctly. Resolves: #816 UserNote: The res_pjsip_config_wizard.so module can now be reloaded. (cherry picked from commit 9e56766cde)	9 months ago
George Joseph	bae7edddcb	voicemail.conf.sample: Fix ':' comment typo ... ...and removed an errant trailing space. Resolves: #819 (cherry picked from commit fa69a286a2)	9 months ago
George Joseph	c77fdb59f4	bridge_softmix: Fix queueing VIDUPDATE control frames ... softmix_bridge_write_control() now calls ast_bridge_queue_everyone_else() with the bridge_channel so the VIDUPDATE control frame isn't echoed back. softmix_bridge_write_control() was setting bridge_channel to NULL when calling ast_bridge_queue_everyone_else() for VIDUPDATE control frames. This was causing the frame to be echoed back to the channel it came from. In certain cases, like when two channels or bridges are being recorded, this can cause a ping-pong effect that floods the system with VIDUPDATE control frames. Resolves: #780 (cherry picked from commit 3be176dfc4)	9 months ago
George Joseph	7a0090325b	manager.c: Add entries to Originate blacklist ... Added Reload and DBdeltree to the list of dialplan application that can't be executed via the Originate manager action without also having write SYSTEM permissions. Added CURL, DB*, FILE, ODBC and REALTIME* to the list of dialplan functions that can't be executed via the Originate manager action without also having write SYSTEM permissions. If the Queue application is attempted to be run by the Originate manager action and an AGI parameter is specified in the app data, it'll be rejected unless the manager user has either the AGI or SYSTEM permissions. Resolves: #GHSA-c4cg-9275-6w44	9 months ago
Asterisk Development Team	c6e587c8a9	Update for certified-18.9-cert10	9 months ago
George Joseph	1fad739c76	.github: Pass app_id and app_priv_key to AsteriskMergePR ... (cherry picked from commit b7a46921c4)	9 months ago
George Joseph	5eff492417	.github: Change OnPRMergeApproved to use default token ... (cherry picked from commit 6cd51f082b)	9 months ago
Sean Bright	d4b70f2194	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. ... Fixes #785 (cherry picked from commit 4ec1be6530)	9 months ago
George Joseph	5c1822a481	app_voicemail_odbc: Allow audio to be kept on disk ... This commit adds a new voicemail.conf option 'odbc_audio_on_disk' which when set causes the ODBC variant of app_voicemail to leave the message and greeting audio files on disk and only store the message metadata in the database. This option came from a concern that the database could grow to large and cause remote access and/or replication to become slow. In a clustering situation with this option, all asterisk instances would share the same database for the metadata and either use a shared filesystem or other filesystem replication service much more suitable for synchronizing files. The changes to app_voicemail to implement this feature were actually quite small but due to the complexity of the module, the actual source code changes were greater. They fall into the following categories: * Tracing. The module is so complex that it was impossible to figure out the path taken for various scenarios without the addition of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in code that's not related to the functional change. Making this worse was the fact that many "if" statements in this module didn't use braces. Since the tracing macros add multiple statements, many "if" statements had to be converted to use braces. * Excessive use of PATH_MAX. Previous maintainers of this module used PATH_MAX to allocate character arrays for filesystem paths and SQL statements as though they cost nothing. In fact, PATH_MAX is defined as 4096 bytes! Some functions had (and still have) multiples of these. One function has 7. Given that the vast majority of installations use the default spool directory path `/var/spool/asterisk/voicemail`, the actual path length is usually less than 80 bytes. That's over 4000 bytes wasted. It was the same for SQL statement buffers. A 4K buffer for statement that only needed 60 bytes. All of these PATH_MAX allocations in the ODBC related code were changed to dynamically allocated buffers. The rest will have to be addressed separately. * Bug fixes. During the development of this feature, several pre-existing ODBC related bugs were discovered and fixed. They had to do with leaving orphaned files on disk, not preserving original message ids when moving messages between folders, not honoring the "formats" config parameter in certain circumstances, etc. UserNote: This commit adds a new voicemail.conf option 'odbc_audio_on_disk' which when set causes the ODBC variant of app_voicemail_odbc to leave the message and greeting audio files on disk and only store the message metadata in the database. Much more information can be found in the voicemail.conf.sample file. (cherry picked from commit c71fbca918)	9 months ago
George Joseph	0203bc0df5	logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT ... If you're tracing a large function that may call another function multiple times in different circumstances, it can be difficult to see from the trace output exactly which location that function was called from. There's no good way to automatically determine the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT simply print out a trace line before and after the call. The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is that SCOPE_CALL ignores the function's return value (if any) where SCOPE_CALL_WITH_RESULT allows you to specify the type of the function's return value so it can be assigned to a variable. SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT and the "int" return type. (cherry picked from commit 741b3c0dbb)	9 months ago
Sean Bright	c543090c81	app_voicemail.c: Completely resequence mailbox folders. ... Resequencing is a process that occurs when we open a voicemail folder and discover that there are gaps between messages (e.g. `msg0000.txt` is missing but `msg0001.txt` exists). Resequencing involves shifting the existing messages down so we end up with a sequential list of messages. Currently, this process stops after reaching a threshold based on the message limit (`maxmsg`) configured on the current folder. However, if `maxmsg` is lowered when a voicemail folder contains more than `maxmsg + 10` messages, resequencing will not run completely leaving the mailbox in an inconsistent state. We now resequence up to the maximum number of messages permitted by `app_voicemail` (currently hard-coded at 9999 messages). Fixes #86 (cherry picked from commit f8f23c6cef)	9 months ago
George Joseph	c912fd2b1a	.github: Use ASTERISKTEAM_PAT for PR merging ... (cherry picked from commit 9945c25f31)	9 months ago
George Joseph	ffdb3d748f	.github: Replace PR workflows with stubs that call reusables ... The PR workflows now are just stubs that call reusable workflows located in the asterisk-ci-actions repo. (cherry picked from commit 0527ae4166)	9 months ago
George Joseph	d5dc0551ad	.github: Refactor NightlyTests to use workflow in asterisk-ci-actions ... (cherry picked from commit a800a20308)	9 months ago
George Joseph	107f992e00	.github: Add branches to workflow_dispatch for NightlyTests ... (cherry picked from commit 1406d558dc)	9 months ago
Asterisk Development Team	407989fa7b	Update for certified-18.9-cert9	12 months ago
Ivan Poddubny	963872daa7	asterisk.c: Fix sending incorrect messages to systemd notify ... Send "RELOADING=1" instead of "RELOAD=1" to follow the format expected by systemd (see sd_notify(3) man page). Do not send STOPPING=1 in remote console mode: attempting to execute "asterisk -rx" by the main process leads to a warning if NotifyAccess=main (the default) or to a forced termination if NotifyAccess=all. (cherry picked from commit 7314a411a9)	12 months ago
Sean Bright	72db4a9530	res_http_websocket.c: Set hostname on client for certificate validation. ... Additionally add a `assert()` to in the TLS client setup code to ensure that hostname is set when it is supposed to be. Fixes #433 (cherry picked from commit 178b2df38a)	12 months ago
George Joseph	10c5aa65e8	tcptls/iostream: Add support for setting SNI on client TLS connections ... If the hostname field of the ast_tcptls_session_args structure is set (which it is for websocket client connections), that hostname will now automatically be used in an SNI TLS extension in the client hello. Resolves: #713 UserNote: Secure websocket client connections now send SNI in the TLS client hello. (cherry picked from commit 7223dfe244)	12 months ago
George Joseph	d1cde2a864	make_buildopts_h: Always include DETECT_DEADLOCKS ... Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must always be included in buildopts.h instead of only when ADD_CFLAGS_TO_BUILDOPTS_H is defined. A SEGV will result otherwise. Resolves: #719 (cherry picked from commit 558d0a8033)	12 months ago
George Joseph	76f922cd1f	rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup ... rtp_engine.c and stun.c were calling ast_register_cleanup which is skipped if any loadable module can't be cleanly unloaded when asterisk shuts down. Since this will always be the case, their cleanup functions never get run. In a practical sense this makes no difference since asterisk is shutting down but if you're in development mode and trying to use the leak sanitizer, the leaks from both of those modules clutter up the output. (cherry picked from commit 00940e2abd)	12 months ago
George Joseph	5e46323179	manager.c: Add missing parameters to Login documentation ... * Added the AuthType and Key parameters for MD5 authentication. * Added the Events parameter. Resolves: #689 (cherry picked from commit d214c72c2a)	12 months ago
George Joseph	4dcdded687	Fix incorrect application and function documentation references ... There were a few references in the embedded documentation XML where the case didn't match or where the referenced app or function simply didn't exist any more. These were causing 404 responses in docs.asterisk.org. (cherry picked from commit 0e0a56a0ca)	12 months ago
George Joseph	06e319b15a	.github: Add PAT to PRSubmitActions/Add Reviewers ... (cherry picked from commit a7e4c193fc)	12 months ago
George Joseph	b11158b852	.github: Remove timeout-minutes from gatetests ... (cherry picked from commit 18f3c355f7)	12 months ago
George Joseph	f7fe5a7ac3	.github: Pass only single GATETEST_COMMAND to AsteriskGateComposite ... (cherry picked from commit f841ae8e62)	12 months ago
Mark Murawski	84bf0a4d17	Remove files that are no longer updated ... Fixes: #360 (cherry picked from commit 81c400a1c2)	12 months ago
Asterisk Development Team	949533f9a9	Update for certified-18.9-cert8	1 year ago
Asterisk Development Team	1ae51f1e1a	Update for certified-18.9-cert8-rc2	1 year ago
George Joseph	ca897dd37f	Rename dialplan_functions.xml to dialplan_functions_doc.xml ... When using COMPILE_DOUBLE, dialplan_functions.xml is mistaken for the source for an embedded XML document and gets compiled to dialplan_functions.o. This causes dialplan_functions.c to be ignored making its functions unavailable and causing chan_pjsip to fail to load.	1 year ago
Sean Bright	c8d4ad1286	openssl: Supress deprecation warnings from OpenSSL 3.0 ... There is work going on to update our OpenSSL usage to avoid the deprecated functions but in the meantime make it possible to compile in devmode. Change-Id: Ib082eb8b3751f0185d8aa8fe127da664c93f0726	1 year ago
Asterisk Development Team	378e010902	Update for certified-18.9-cert8-rc1	1 year ago
George Joseph	8735277a4a	.github: Add force_cherry_pick option to Releaser ... (cherry picked from commit 031703c596)	1 year ago
George Joseph	5988bb529e	.github: Remove start_version from Releaser ... (cherry picked from commit 7d65b0ad80)	1 year ago
Mike Bradeen	33e95dfb8b	app_chanspy: Add 'D' option for dual-channel audio ... Adds the 'D' option to app chanspy that causes the input and output frames of the spied channel to be interleaved in the spy output frame. This allows the input and output of the spied channel to be decoded separately by the receiver. If the 'o' option is also set, the 'D' option is ignored as the audio being spied is inherently one direction. Fixes: #569 UserNote: The ChanSpy application now accepts the 'D' option which will interleave the spied audio within the outgoing frames. The purpose of this is to allow the audio to be read as a Dual channel stream with separate incoming and outgoing audio. Setting both the 'o' option and the 'D' option and results in the 'D' option being ignored. (cherry picked from commit e0be8be337)	1 year ago
George Joseph	6d29836274	.github: Update github-script to v7 and fix a rest bug ... Need to update the github-script to v7 to squash deprecation warnings. Also fixed the API name for github.rest.pulls.requestReviewers. (cherry picked from commit be9b5bf0a9)	1 year ago
Naveen Albert	a26faa9ac5	manager.c: Fix regression due to using wrong free function. ... Commit 424be34563 introduced a regression by calling ast_free on memory allocated by realpath. This causes Asterisk to abort when executing this function. Since the memory is allocated by glibc, it should be freed using ast_std_free. Resolves: #513 (cherry picked from commit 3ff081e581)	1 year ago