33323 Commits (fa69a286a2dda7eb01ca27ead577fc5a1a3ffcef)

 

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Author	SHA1	Message	Date
George Joseph	fa69a286a2	voicemail.conf.sample: Fix ':' comment typo ... ...and removed an errant trailing space. Resolves: #819	1 year ago
George Joseph	3be176dfc4	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	2 years ago
George Joseph	b7a46921c4	.github: Pass app_id and app_priv_key to AsteriskMergePR	2 years ago
George Joseph	6cd51f082b	.github: Change OnPRMergeApproved to use default token	2 years ago
Sean Bright	4ec1be6530	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. ... Fixes #785	2 years ago
George Joseph	c71fbca918	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.	2 years ago
George Joseph	741b3c0dbb	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.	2 years ago
Sean Bright	f8f23c6cef	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	2 years ago
George Joseph	9945c25f31	.github: Use ASTERISKTEAM_PAT for PR merging	2 years ago
George Joseph	0527ae4166	.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.	2 years ago
George Joseph	a800a20308	.github: Refactor NightlyTests to use workflow in asterisk-ci-actions	2 years ago
George Joseph	1406d558dc	.github: Add branches to workflow_dispatch for NightlyTests	2 years ago
Ivan Poddubny	7314a411a9	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.	2 years ago
Sean Bright	178b2df38a	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	2 years ago
George Joseph	7223dfe244	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.	2 years ago
George Joseph	558d0a8033	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	2 years ago
George Joseph	00940e2abd	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.	2 years ago
George Joseph	d214c72c2a	manager.c: Add missing parameters to Login documentation ... * Added the AuthType and Key parameters for MD5 authentication. * Added the Events parameter. Resolves: #689	2 years ago
George Joseph	0e0a56a0ca	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.	2 years ago
George Joseph	a7e4c193fc	.github: Add PAT to PRSubmitActions/Add Reviewers	2 years ago
George Joseph	18f3c355f7	.github: Remove timeout-minutes from gatetests	2 years ago
George Joseph	f841ae8e62	.github: Pass only single GATETEST_COMMAND to AsteriskGateComposite	2 years ago
George Joseph	386cdef149	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.	2 years ago
Sean Bright	b4ecf5eaf1	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	2 years ago
George Joseph	031703c596	.github: Add force_cherry_pick option to Releaser	2 years ago
George Joseph	7d65b0ad80	.github: Remove start_version from Releaser	2 years ago
Mike Bradeen	e0be8be337	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.	2 years ago
George Joseph	be9b5bf0a9	.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.	2 years ago
Naveen Albert	3ff081e581	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	2 years ago
George Joseph	a4f9d885a7	res_rtp_asterisk: Fix regression issues with DTLS client check ... * Since ICE candidates are used for the check and pjproject is required to use ICE, res_rtp_asterisk was failing to compile when pjproject wasn't available. The check is now wrapped with an #ifdef HAVE_PJPROJECT. * The rtp->ice_active_remote_candidates container was being used to check the address on incoming packets but that container doesn't contain peer reflexive candidates discovered during negotiation. This was causing the check to fail where it shouldn't. We now check against pjproject's real_ice->rcand array which will contain those candidates. * Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where we weren't zeroing out sin->sin_zero before returning. This was causing ast_sockaddr_cmp() to always return false when one of the inputs was converted from a pj_sockaddr, even if both inputs had the same address and port. Resolves: #500 Resolves: #503 Resolves: #505	2 years ago
George Joseph	85fc4ce712	doc: Remove obsolete CHANGES-staging directrory ... This should have been removed after the last release but was missed.	2 years ago
Gitea	6b4f9ab005	res_pjsip_header_funcs: Duplicate new header value, don't copy. ... When updating an existing header the 'update' code incorrectly just copied the new value into the existing buffer. If the new value exceeded the available buffer size memory outside of the buffer would be written into, potentially causing a crash. This change makes it so that the 'update' now duplicates the new header value instead of copying it into the existing buffer.	2 years ago
George Joseph	2b66362173	res_rtp_asterisk.c: Check DTLS packets against ICE candidate list ... When ICE is in use, we can prevent a possible DOS attack by allowing DTLS protocol messages (client hello, etc) only from sources that are in the active remote candidates list. Resolves: GHSA-hxj9-xwr8-w8pq	2 years ago
Ben Ford	705cd2845d	manager.c: Prevent path traversal with GetConfig. ... When using AMI GetConfig, it was possible to access files outside of the Asterisk configuration directory by using filenames with ".." and "./" even while live_dangerously was not enabled. This change resolves the full path and ensures we are still in the configuration directory before attempting to access the file.	2 years ago
Mike Bradeen	c7050787f3	res_pjsip: disable raw bad packet logging ... Add patch to split the log level for invalid packets received on the signaling port. The warning regarding the packet will move to level 2 so that it can still be displayed, while the raw packet will be at level 4.	2 years ago
George Joseph	0f20f39db8	MergeApproved.yml: Remove unneeded concurrency ... The concurrency parameter on the MergeAndCherryPick job has been rmeoved. It was a hold-over from earlier days.	2 years ago
George Joseph	c53cd1c82d	SECURITY.md: Update with correct documentation URL	2 years ago
George Joseph	8e012faf9e	chan_pjsip: Add PJSIPHangup dialplan app and manager action ... See UserNote below. Exposed the existing Hangup AMI action in manager.c so we can use all of it's channel search and AMI protocol handling without duplicating that code in dialplan_functions.c. Added a lookup function to res_pjsip.c that takes in the string represenation of the pjsip_status_code enum and returns the actual status code. I.E. ast_sip_str2rc("DECLINE") returns 603. This allows the caller to specify PJSIPHangup(decline) in the dialplan, just like Hangup(call_rejected). Also extracted the XML documentation to its own file since it was almost as large as the code itself. UserNote: A new dialplan app PJSIPHangup and AMI action allows you to hang up an unanswered incoming PJSIP call with a specific SIP response code in the 400 -> 699 range.	2 years ago
Mark Murawski	81c400a1c2	Remove files that are no longer updated ... Fixes: #360	2 years ago
Mike Bradeen	db5767f19d	res_speech: allow speech to translate input channel ... * Allow res_speech to translate the input channel if the format is translatable to a format suppored by the speech provider. Resolves: #129 UserNote: res_speech now supports translation of an input channel to a format supported by the speech provider, provided a translation path is available between the source format and provider capabilites.	2 years ago
George Joseph	05d26994af	.github: PRSubmitActions: Fix adding reviewers to PR	2 years ago
George Joseph	cd640dc67b	.github: New PR Submit workflows ... The workflows that get triggered when PRs are submitted or updated have been replaced with ones that are more secure and have a higher level of parallelism.	2 years ago
George Joseph	5eb676bab5	.github: New PR Submit workflows ... The workflows that get triggered when PRs are submitted or updated have been replaced with ones that are more secure and have a higher level of parallelism.	2 years ago
Mike Bradeen	5342562892	res_stasis: signal when new command is queued ... res_statsis's app loop sleeps for up to .2s waiting on input to a channel before re-checking the command queue. This can cause delays between channel setup and bridge. This change is to send a SIGURG on the sleeping thread when a new command is enqueued. This exits the sleeping thread out of the ast_waitfor() call triggering the new command being processed on the channel immediately. Resolves: #362 UserNote: Call setup times should be significantly improved when using ARI.	2 years ago
George Joseph	20b06e7bb5	logger.h: Add ability to change the prefix on SCOPE_TRACE output ... You can now define the _TRACE_PREFIX_ macro to change the default trace line prefix of "file:line function" to something else. Full documentation in logger.h.	2 years ago
George Joseph	d0a1c2b026	.github: Fix job prereqs in PROpenedUpdated	2 years ago
George Joseph	825d3b981a	.github: Block PR tests until approved	2 years ago
George Joseph	733ab35848	Add libjwt to third-party ... The current STIR/SHAKEN implementation is not currently usable due to encryption issues. Rather than trying to futz with OpenSSL and the the current code, we can take advantage of the existing capabilities of libjwt but we first need to add it to the third-party infrastructure already in place for jansson and pjproject. A few tweaks were also made to the third-party infrastructure as a whole. The jansson "dest" install directory was renamed "dist" to better match convention, and the third-party Makefile was updated to clean all product directories not just the ones currently in use. Resolves: #349	2 years ago
Mike Bradeen	7369de127a	res_pjsip: update qualify_timeout documentation with DNS note ... The documentation on qualify_timeout does not explicitly state that the timeout includes any time required to perform any needed DNS queries on the endpoint. If the OPTIONS response is delayed due to the DNS query, it can still render an endpoint as Unreachable if the net time is enough for qualify_timeout to expire. Resolves: #352	2 years ago
George Joseph	04183eb246	lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS ... Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS. Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just causes the lock calls to loop over trylock in 200us intervals until the lock is obtained and spits out log messages if it takes more than 5 seconds. From a code perspective, the only reason they were tied together was for logging. So... The ifdefs in lock.c were refactored to allow DETECT_DEADLOCKS to be enabled without also enabling DEBUG_THREADS. Resolves: #321 UserNote: You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS. This removes a significant amount of overhead if you just want to detect possible deadlocks vs needing full lock tracing.	2 years ago