asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	f55df3621a	logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds. Fixes #785	1 year ago
George Joseph	a9b74ffa52	.github: Use ASTERISKTEAM_PAT for PR merging	1 year ago
George Joseph	27f7cb6ea0	stasis_channels: Use uniqueid and name to delete old snapshots Whenver a new channel snapshot is created or when a channel is destroyed, we need to delete any existing channel snapshot from the snapshot cache. Historically, we used the channel->snapshot pointer to delete any existing snapshots but this has two issues. First, if something (possibly ast_channel_internal_swap_snapshots) sets channel->snapshot to NULL while there's still a snapshot in the cache, we wouldn't be able to delete it and it would be orphaned when the channel is destroyed. Since we use the cache to list channels from the CLI, AMI and ARI, it would appear as though the channel was still there when it wasn't. Second, since there are actually two caches, one indexed by the channel's uniqueid, and another indexed by the channel's name, deleting from the caches by pointer requires a sequential search of all of the hash table buckets in BOTH caches to find the matching snapshots. Not very efficient. So, we now delete from the caches using the channel's uniqueid and name. This solves both issues. This doesn't address how channel->snapshot might have been set to NULL in the first place because although we have concrete evidence that it's happening, we haven't been able to reproduce it. Resolves: #783	1 year ago
George Joseph	b133383389	.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.	1 year ago
George Joseph	b867fa2fe5	.github: Refactor NightlyTests to use workflow in asterisk-ci-actions	1 year ago
George Joseph	1b3a73cb24	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.	1 year ago
Tinet-mucw	68a9c5683a	bridge_basic.c: Make sure that ast_bridge_channel is not destroyed while iterating over bridge->channels. From the gdb information, we can see that while iterating over bridge->channels, the ast_bridge_channel reference count is 0, indicating it has already been destroyed.Additionally, when ast_bridge_channel is removed from bridge->channels, the bridge is first locked. Therefore, locking the bridge before iterating over bridge->channels can resolve the race condition. Resolves: https://github.com/asterisk/asterisk/issues/768	1 year ago
Alexei Gradinari	192a848311	app_queue: Add option to not log Restricted Caller ID to queue_log Add a queue option log-restricted-caller-id to strip the Caller ID when storing the ENTERQUEUE event in the queue log if the Caller ID is restricted. Resolves: #765 UpgradeNote: Add a new column to the queues table: queue_log_option_log_restricted ENUM('0','1','off','on','false','true','no','yes') to control whether the Restricted Caller ID will be stored in the queue log. UserNote: Add a Queue option log-restricted-caller-id to control whether the Restricted Caller ID will be stored in the queue log. If log-restricted-caller-id=no then the Caller ID will be stripped if the Caller ID is restricted.	1 year ago
Alexei Gradinari	8d7ee89047	pbx.c: expand fields width of "core show hints" The current width for "extension" is 20 and "device state id" is 20, which is too small. The "extension" field contains "ext"@"context", so 20 characters is not enough. The "device state id" field, for example for Queue pause state contains Queue:"queue_name"_pause_PSJIP/"endpoint", so the 20 characters is not enough. Increase the width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters. Resolves: #770 UserNote: The fields width of "core show hints" were increased. The width of "extension" field to 30 characters and the width of the "device state id" field to 60 characters.	1 year ago
Sean Bright	78d63bc11c	pjsip: Add PJSIP_PARSE_URI_FROM dialplan function. Various SIP headers permit a URI to be prefaced with a `display-name` production that can include characters (like commas and parentheses) that are problematic for Asterisk's dialplan parser and, specifically in the case of this patch, the PJSIP_PARSE_URI function. This patch introduces a new function - `PJSIP_PARSE_URI_FROM` - that behaves identically to `PJSIP_PARSE_URI` except that the first argument is now a variable name and not a literal URI. Fixes #756	1 year ago
Sean Bright	7920e75da5	manager.c: Properly terminate `CoreShowChannelMap` event. Fixes #761	1 year ago
Bastian Triller	b56d50ba16	cli: Show configured cache dir Since Asterisk 19 it is possible to cache recorded files into another directory [1] [2]. Show configured location of cache dir in CLI's core show settings. [1] ASTERISK-29143 [2] `b08427134f`	1 year ago
Sean Bright	f9a359c5c5	xml.c: Update deprecated libxml2 API usage. Two functions are deprecated as of libxml2 2.12: * xmlSubstituteEntitiesDefault * xmlParseMemory So we update those with supported API. Additionally, `res_calendar_caldav` has been updated to use libxml2's xmlreader API instead of the SAX2 API which has always felt a little hacky (see deleted comment block in `res_calendar_caldav.c`). The xmlreader API has been around since libxml2 2.5.0 which was released in 2003. Fixes #725	1 year ago
chrsmj	735330bbd1	cdr_pgsql: Fix crash when the module fails to load multiple times. Missing or corrupt cdr_pgsql.conf configuration file can cause the second attempt to load the PostgreSQL CDR module to crash Asterisk via the Command Line Interface because a null CLI command is registered on the first failed attempt to load the module. Resolves: #736	1 year ago
Sean Bright	fa043c8239	asterisk.c: Don't log an error if .asterisk_history does not exist. Fixes #751	1 year ago
Walter Doekes	f5a6ff5de5	chan_ooh323: Fix R/0 typo in docs Change-Id: I00d79a807dd0fea9e42fca919aa2fb0b71cdf20e	1 year ago
Sean Bright	542cd37830	bundled_pjproject: Disable UPnP support. Fixes #747	1 year ago
Sean Bright	d51a86dc91	file.h: Rename function argument to avoid C++ keyword clash. Fixes #744	1 year ago
George Joseph	c014b66554	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	1 year ago
George Joseph	12b997312b	.github: Add branches to workflow_dispatch for NightlyTests	1 year ago
Mike Bradeen	624f509ce4	rtp_engine: add support for multirate RFC2833 digits Add RFC2833 DTMF support for 16K, 24K, and 32K bitrate codecs. Asterisk currently treats RFC2833 Digits as a single rtp payload type with a fixed bitrate of 8K. This change would expand that to 8, 16, 24 and 32K. This requires checking the offered rtp types for any of these bitrates and then adding an offer for each (if configured for RFC2833.) DTMF generation must also be changed in order to look at the current outbound codec in order to generate appropriately timed rtp. For cases where no outgoing audio has yet been sent prior to digit generation, Asterisk now has a concept of a 'preferred' codec based on offer order. On inbound calls Asterisk will mimic the payload types of the RFC2833 digits. On outbound calls Asterisk will choose the next free payload types starting with 101. UserNote: No change in configuration is required in order to enable this feature. Endpoints configured to use RFC2833 will automatically have this enabled. If the endpoint does not support this, it should not include it in the SDP offer/response. Resolves: #699	1 year ago
Ivan Poddubny	685f525b28	configs: Fix a misleading IPv6 ACL example in Named ACLs "deny=::" is equivalent to "::/128". In order to mean "deny everything by default" it must be "::/0".	1 year ago
Ivan Poddubny	d3ff7c3eee	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.	1 year ago
Fabrice Fontaine	c47307567a	res/stasis/control.c: include signal.h Include signal.h to avoid the following build failure with uclibc-ng raised since `2694792e13`: stasis/control.c: In function 'exec_command_on_condition': stasis/control.c:313:3: warning: implicit declaration of function 'pthread_kill'; did you mean 'pthread_yield'? [-Wimplicit-function-declaration] 313 \| pthread_kill(control->control_thread, SIGURG); \| ^~~~~~~~~~~~ \| pthread_yield stasis/control.c:313:41: error: 'SIGURG' undeclared (first use in this function) 313 \| pthread_kill(control->control_thread, SIGURG); \| ^~~~~~ cherry-pick-to: 18 cherry-pick-to: 20 cherry-pick-to: 21 Fixes: #729	1 year ago
Naveen Albert	9fc596aaa7	res_pjsip_logger: Preserve logging state on reloads. Currently, reloading res_pjsip will cause logging to be disabled. This is because logging can also be controlled via the debug option in pjsip.conf and this defaults to "no". To improve this, logging is no longer disabled on reloads if logging had not been previously enabled using the debug option from the config. This ensures that logging enabled from the CLI will persist through a reload. ASTERISK-29912 #close Resolves: #246 UserNote: Issuing "pjsip reload" will no longer disable logging if it was previously enabled from the CLI.	1 year ago
Naveen Albert	57bb09667d	logger: Add unique verbose prefixes for levels 5-10. Add unique verbose prefixes for levels higher than 4, so that these can be visually differentiated from each other. Resolves: #721	1 year ago
Naveen Albert	3a49f9ade5	say.c: Fix cents off-by-one due to floating point rounding. Some of the money announcements can be off by one cent, due to the use of floating point in the money calculations, which is bad for obvious reasons. This replaces floating point with simple string parsing to ensure the cents value is converted accurately. Resolves: #525	1 year ago
Naveen Albert	a056e94885	loader.c: Allow dependent modules to be unloaded recursively. Because of the (often recursive) nature of module dependencies in Asterisk, hot swapping a module on the fly is cumbersome if a module is depended on by other modules. Currently, dependencies must be popped manually by unloading dependents, unloading the module of interest, and then loading modules again in reverse order. To make this easier, the ability to do this recursively in certain circumstances has been added, as an optional extension to the "module refresh" command. If requested, Asterisk will check if a module that has a positive usecount could be unloaded safely if anything recursively dependent on it were unloaded. If so, it will go ahead and unload all these modules and load them back again. This makes hot swapping modules that provide dependencies much easier. Resolves: #474 UserNote: In certain circumstances, modules with dependency relations can have their dependents automatically recursively unloaded and loaded again using the "module refresh" CLI command or the ModuleLoad AMI command.	1 year ago
Henrik Liljedahl	1423cfee29	res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeout setting. First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation. Fixes #710	1 year ago
George Joseph	4d6f84a14f	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.	1 year ago
George Joseph	b7ed77a7c5	stir_shaken: Fix memory leak, typo in config, tn canonicalization * Fixed possible memory leak in tn_config:tn_get_etn() where we weren't releasing etn if tn or eprofile were null. * We now canonicalize TNs before using them for lookups or adding them to Identity headers. * Fixed a typo in stir_shaken.conf.sample. Resolves: #716	1 year ago
George Joseph	d1d80af0c0	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	1 year ago
Spiridonov Dmitry	b8525e87ae	sorcery.c: Fixed crash error when executing "module reload" Fixed crash error when cli "module reload". The error appears when compiling with res_prometheus and using the sorcery memory cache for registrations	1 year ago
Naveen Albert	4cf8d9d94a	callerid.c: Parse previously ignored Caller ID parameters. Commit `f2f397c1a8` previously made it possible to send Caller ID parameters to FXS stations which, prior to that, could not be sent. This change is complementary in that we now handle receiving all these parameters on FXO lines and provide these up to the dialplan, via chan_dahdi. In particular: * If a redirecting reason is provided, the channel's redirecting reason is set. No redirecting number is set, since there is no parameter for this in the Caller ID protocol, but the reason can be checked to determine if and why a call was forwarded. * If the Call Qualifier parameter is received, the Call Qualifier variable is set. * Some comments have been added to explain why some of the code is the way it is, to assist other people looking at it. With this change, Asterisk's Caller ID implementation is now reasonably complete for both FXS and FXO operation. Resolves: #681	1 year ago
George Joseph	cf5c46d8ae	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.	1 year ago
Sean Bright	6914c93791	app_queue.c: Properly handle invalid strategies from realtime. The existing code sets the queue strategy to `ringall` but it is then immediately overwritten with an invalid one. Fixes #707	1 year ago
Naveen Albert	051ddbe729	file.c, channel.c: Don't emit warnings if progress received. Silently ignore AST_CONTROL_PROGRESS where appropriate, as most control frames already are. Resolves: #696	1 year ago
Sean Bright	df6f30022f	alembic: Correct NULLability of PJSIP id columns. Fixes #695	2 years ago
George Joseph	1eec3b6d18	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	b2bcc0cd09	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
Naveen Albert	9ba6875a3f	func_callerid: Emit warning if invalid redirecting reason set. Emit a warning if REDIRECTING(reason) is set to an invalid reason, consistent with what happens when REDIRECTING(orig-reason) is set to an invalid reason. Resolves: #683	2 years ago
Naveen Albert	4ff93474df	chan_dahdi: Add DAHDIShowStatus AMI action. * Add an AMI action to correspond to the "dahdi show status" command, allowing span information to be retrieved via AMI. * Show span number and sig type in "dahdi show channels". Resolves: #673	2 years ago
Sperl Viktor	c8769f3d5a	res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address. Add a new identify_by option to res_pjsip_endpoint_identifier_ip called 'transport' this matches endpoints based on the bound ip address (local) instead of the 'ip' option, which matches on the source ip address (remote). UserNote: set identify_by=transport for the pjsip endpoint. Then use the existing 'match' option and the new 'transport' option of the identify. Fixes: #672	2 years ago
George Joseph	16b264d6a9	res_stir_shaken: Fix compilation for CentOS7 (openssl 1.0.2) * OpenSSL 1.0.2 doesn't support X509_get0_pubkey so we now use X509_get_pubkey. The difference is that X509_get_pubkey requires the caller to free the EVP_PKEY themselves so we now let RAII_VAR do that. * OpenSSL 1.0.2 doesn't support upreffing an X509_STORE so we now wrap it in an ao2 object. * OpenSSL 1.0.2 doesn't support X509_STORE_get0_objects to get all the certs from an X509_STORE and there's no easy way to polyfill it so the CLI commands that list profiles will show a "not supported" message instead of listing the certs in a store. Resolves: #676	2 years ago
George Joseph	9e2179baa1	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
Sean Bright	41daf01866	cli.c: `core show channels concise` is not really deprecated. Fixes #675	2 years ago
Sperl Viktor	ac297d15f8	res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI Add ability to match against PJSIP request URI. UserNote: this new feature let users match endpoints based on the indound SIP requests' URI. To do so, add 'request_uri' to the endpoint's 'identify_by' option. The 'match_request_uri' option of the identify can be an exact match for the entire request uri, or a regular expression (between slashes). It's quite similar to the header identifer. Fixes: #599	2 years ago
Naveen Albert	63aa08fa0b	chan_dahdi: Don't retry opening nonexistent channels on restart. Commit `729cb1d390` added logic to retry opening DAHDI channels on "dahdi restart" if they failed initially, up to 1,000 times in a loop, to address cases where the channel was still in use. However, this retry loop does not use the actual error, which means chan_dahdi will also retry opening nonexistent channels 1,000 times per channel, causing a flood of unnecessary warning logs for an operation that will never succeed, with tens or hundreds of thousands of open attempts being made. The original patch would have been more targeted if it only retried on the specific relevant error (likely EBUSY, although it's hard to say since the original issue is no longer available). To avoid the problem above while avoiding the possibility of breakage, this skips the retry logic if the error is ENXIO (No such device or address), since this will never succeed. Resolves: #669	2 years ago
Joshua Elson	555eb9d3d2	Implement Configurable TCP Keepalive Settings in PJSIP Transports This commit introduces configurable TCP keepalive settings for both TCP and TLS transports. The changes allow for finer control over TCP connection keepalives, enhancing stability and reliability in environments prone to connection timeouts or where intermediate devices may prematurely close idle connections. This has proven necessary and has already been tested in production in several specialized environments where access to the underlying transport is unreliable in ways invisible to the operating system directly, so these keepalive and timeout mechanisms are necessary. Fixes #657	2 years ago
Martin Tomec	cba82273ae	res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA There was functionality in chan_sip to get REFER headers, with GET_TRANSFERRER_DATA variable. This commit implements the same functionality in pjsip, to ease transfer from chan_sip to pjsip. Fixes: #579 UserNote: the GET_TRANSFERRER_DATA dialplan variable can now be used also in pjsip.	2 years ago

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