33943 Commits (8a20121298386281c3d5d59bdb6236991ccf01cf)

 

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Author	SHA1	Message	Date
Tinet-mucw	c759489d3e	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 (cherry picked from commit a1d0dac6c6)	1 year ago
Alexei Gradinari	1108f61bfb	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. (cherry picked from commit 624c23310e)	1 year ago
Alexei Gradinari	e93c5e945a	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. (cherry picked from commit dcdda4688e)	1 year ago
Sean Bright	d18159a008	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 (cherry picked from commit 5068cc814f)	1 year ago
Sean Bright	9afc995824	manager.c: Properly terminate `CoreShowChannelMap` event. ... Fixes #761 (cherry picked from commit 87278c087a)	1 year ago
Sean Bright	a2980fe5fa	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 (cherry picked from commit 126cb5a20d)	1 year ago
chrsmj	9f369fb5fd	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 (cherry picked from commit 4fea8a433b)	1 year ago
Sean Bright	877d5ff577	asterisk.c: Don't log an error if .asterisk_history does not exist. ... Fixes #751 (cherry picked from commit e9fb27f804)	1 year ago
Walter Doekes	823c948842	chan_ooh323: Fix R/0 typo in docs ... Change-Id: I00d79a807dd0fea9e42fca919aa2fb0b71cdf20e (cherry picked from commit 4f998eb5aa)	1 year ago
Sean Bright	6042ce4a78	bundled_pjproject: Disable UPnP support. ... Fixes #747 (cherry picked from commit ff510c644a)	1 year ago
Sean Bright	f0b48a63b6	file.h: Rename function argument to avoid C++ keyword clash. ... Fixes #744 (cherry picked from commit 9942ebae86)	1 year ago
George Joseph	056d83043e	.github: Add branches to workflow_dispatch for NightlyTests ... (cherry picked from commit d2456e0436)	1 year ago
Mike Bradeen	992681cc45	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 (cherry picked from commit 6bf66b82d7)	1 year ago
Asterisk Development Team	351c871038	Update for 18.23.1	1 year ago
George Joseph	3572cc8f4b	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." ... This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	1 year ago
Asterisk Development Team	14c4343b72	Update for 18.23.0	1 year ago
Asterisk Development Team	ccebe76dcd	Update for 18.23.0-rc1	1 year ago
Ivan Poddubny	5ae0c186aa	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". (cherry picked from commit ee6a0365ab)	1 year ago
Ivan Poddubny	3988aff68e	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 3e9922d389)	1 year ago
Fabrice Fontaine	9efb9c7d45	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 (cherry picked from commit 044f3e85ba)	1 year ago
Naveen Albert	fc35e92ece	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. (cherry picked from commit 3b600e2787)	1 year ago
Naveen Albert	1acf473c5b	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 (cherry picked from commit 23f5ce69fd)	1 year ago
Naveen Albert	62f9c54f4c	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 (cherry picked from commit b836f9c01c)	1 year ago
Naveen Albert	46ddecb3d5	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. (cherry picked from commit c276ae11e0)	1 year ago
Henrik Liljedahl	f580b8b5f1	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 (cherry picked from commit d74266fe05)	1 year ago
George Joseph	229f3035f0	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 f9a1e3095c)	1 year ago
George Joseph	7d1a8e8411	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 (cherry picked from commit 3d2def92e2)	1 year ago
George Joseph	45eec15df1	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 6568c86989)	1 year ago
Spiridonov Dmitry	f817526226	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 (cherry picked from commit 5a883f5410)	1 year ago
Naveen Albert	a2579ec402	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 (cherry picked from commit 6bd0b67081)	1 year ago
George Joseph	6e8678d8ac	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 65018e8ebf)	1 year ago
Sean Bright	e49049b474	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 (cherry picked from commit 574d6986ce)	1 year ago
Naveen Albert	6576f877e5	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 (cherry picked from commit 76b2410143)	1 year ago
Sean Bright	572e6e363e	alembic: Correct NULLability of PJSIP id columns. ... Fixes #695 (cherry picked from commit a69b8cdf42)	1 year ago
George Joseph	ee2213e3a9	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 ce9e3d9275)	1 year ago
George Joseph	5a379bc875	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 579126188d)	1 year ago
Naveen Albert	bc5db3d687	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 (cherry picked from commit 376bfbbb25)	1 year ago
Naveen Albert	6e7f26b795	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 (cherry picked from commit 37eab0cbd6)	1 year ago
Sperl Viktor	c2be3097c6	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 (cherry picked from commit 62bee37d0d)	1 year ago
George Joseph	2fd379d3be	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 (cherry picked from commit 4cb56ccd28)	1 year ago
George Joseph	ca6590426a	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 9c2cc5bf24)	1 year ago
Sean Bright	bfb8019a83	cli.c: `core show channels concise` is not really deprecated. ... Fixes #675 (cherry picked from commit 8f797b883c)	1 year ago
Sperl Viktor	779755527a	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 (cherry picked from commit 83f1317eb4)	1 year ago
Joshua Elson	b85c9d80ca	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 (cherry picked from commit f61e1d902b)	1 year ago
Naveen Albert	5ccedf22d2	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 (cherry picked from commit ec7e6fe2ed)	1 year ago
Martin Tomec	bbd868f62f	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. (cherry picked from commit 7febbed29f)	1 year ago
Martin Nystroem	622cff05c9	res_ari.c: Add additional output to ARI requests when debug is enabled ... When ARI debug is enabled the logs will now output http method and the uri. Fixes: #666 (cherry picked from commit a56d4f7936)	1 year ago
Sean Bright	dd0d4efc7c	alembic: Fix compatibility with SQLAlchemy 2.0+. ... SQLAlchemy 2.0 changed the way that commits/rollbacks are handled causing the final `UPDATE` to our `alembic_version_<whatever>` tables to be rolled back instead of committed. We now use one connection to determine which `alembic_version_<whatever>` table to use and another to run the actual migrations. This prevents the erroneous rollback. This change is compatible with both SQLAlchemy 1.4 and 2.0. (cherry picked from commit 9bdbcf0aad)	1 year ago
jonatascalebe	3705d9e16e	manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action ... manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action The action originate does not has the ability to run an subroutine at initial channel, like the Aplication Originate. This update give this ability for de action originate too. For example, we can run a routine via Gosub on the channel to request an automatic answer, so the caller does not need to accept the call when using the originate command via manager, making the operation more efficient. UserNote: When using the Originate AMI Action, we now can pass the PreDialGoSub parameter, instructing the asterisk to perform an subrouting at channel before call start. With this parameter an call initiated by AMI can request the channel to start the call automaticaly, adding a SIP header to using GoSUB, instructing to autoanswer the channel, and proceeding the outbuound extension executing. Exemple of an context to perform the previus indication: [addautoanswer] exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0) exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0) exten => _s,n,Return() (cherry picked from commit 8513881334)	1 year ago
Naveen Albert	7cce7c9397	menuselect: Minor cosmetic fixes. ... Improve some of the formatting from dd3f17c699 (#521). (cherry picked from commit 31cdf64d85)	1 year ago