asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	ff510c644a	bundled_pjproject: Disable UPnP support. Fixes #747	11 months ago
Sean Bright	9942ebae86	file.h: Rename function argument to avoid C++ keyword clash. Fixes #744	11 months ago
George Joseph	463cf883d9	Revert "res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address." This reverts PR #602 Resolves: #GHSA-qqxj-v78h-hrf9	11 months ago
George Joseph	d2456e0436	.github: Add branches to workflow_dispatch for NightlyTests	12 months ago
Mike Bradeen	6bf66b82d7	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	12 months ago
Ivan Poddubny	ee6a0365ab	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".	12 months ago
Ivan Poddubny	3e9922d389	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.	12 months ago
Fabrice Fontaine	044f3e85ba	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	12 months ago
Naveen Albert	3b600e2787	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.	12 months ago
Naveen Albert	23f5ce69fd	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	b836f9c01c	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	c276ae11e0	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	d74266fe05	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	f9a1e3095c	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	3d2def92e2	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	6568c86989	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	5a883f5410	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	6bd0b67081	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	65018e8ebf	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	574d6986ce	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	76b2410143	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	a69b8cdf42	alembic: Correct NULLability of PJSIP id columns. Fixes #695	1 year ago
George Joseph	ce9e3d9275	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.	1 year ago
George Joseph	579126188d	manager.c: Add missing parameters to Login documentation * Added the AuthType and Key parameters for MD5 authentication. * Added the Events parameter. Resolves: #689	1 year ago
Naveen Albert	376bfbbb25	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	1 year ago
Naveen Albert	37eab0cbd6	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	1 year ago
Sperl Viktor	62bee37d0d	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	1 year ago
George Joseph	4cb56ccd28	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	1 year ago
George Joseph	9c2cc5bf24	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.	1 year ago
Sean Bright	8f797b883c	cli.c: `core show channels concise` is not really deprecated. Fixes #675	1 year ago
Sperl Viktor	83f1317eb4	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	1 year ago
Joshua Elson	f61e1d902b	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	1 year ago
Naveen Albert	ec7e6fe2ed	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	1 year ago
Martin Tomec	7febbed29f	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.	1 year ago
Martin Nystroem	a56d4f7936	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	1 year ago
Sean Bright	9bdbcf0aad	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.	1 year ago
jonatascalebe	8513881334	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()	1 year ago
Naveen Albert	31cdf64d85	menuselect: Minor cosmetic fixes. Improve some of the formatting from `dd3f17c699` (#521).	1 year ago
Naveen Albert	264ffaebb1	pbx_variables.c: Prevent SEGV due to stack overflow. It is possible for dialplan to result in an infinite recursion of variable substitution, which eventually leads to stack overflow. If we detect this, abort substitution and log an error for the user to fix the broken dialplan. Resolves: #480 UpgradeNote: The maximum amount of dialplan recursion using variable substitution (such as by using EVAL_EXTEN) is capped at 15.	1 year ago
Holger Hans Peter Freyther	7b1807413d	res_prometheus: Fix duplicate output of metric and help text The prometheus exposition format requires each line to be unique[1]. This is handled by struct prometheus_metric having a list of children that is managed when registering a metric. In case the scrape callback is used, it is the responsibility of the implementation to handle this correctly. Originally the bridge callback didn't handle NULL snapshots, the crash fix lead to NULL metrics, and fixing that lead to duplicates. The original code assumed that snapshots are not NULL and then relied on "if (i > 0)" to establish the parent/children relationship between metrics of the same class. This is not workerable as the first bridge might be invisible/lacks a snapshot. Fix this by keeping a separate array of the first metric by class. Instead of relying on the index of the bridge, check whether the array has an entry. Use that array for the output. Add a test case that verifies that the help text is not duplicated. Resolves: #642 [1] https://prometheus.io/docs/instrumenting/exposition_formats/#grouping-and-sorting	1 year ago
Naveen Albert	6f5ca2cb51	manager.c: Add CLI command to kick AMI sessions. This adds a CLI command that can be used to manually kick specific AMI sessions. Resolves: #485 UserNote: The "manager kick session" CLI command now allows kicking a specified AMI session.	1 year ago
Naveen Albert	e10bca9f58	chan_dahdi: Allow specifying waitfordialtone per call. The existing "waitfordialtone" setting in chan_dahdi.conf applies permanently to a specific channel, regardless of how it is being used. This rather restrictively prevents a system from simultaneously being able to pick free lines for outgoing calls while also allowing barge-in to a trunk by some other arrangement. This allows specifying "waitfordialtone" using the CHANNEL function for only the next call that will be placed, allowing significantly more flexibility in the use of trunk interfaces. Resolves: #472 UserNote: "waitfordialtone" may now be specified for DAHDI trunk channels on a per-call basis using the CHANNEL function.	1 year ago
Naveen Albert	d0b51f1ff7	res_parking: Fail gracefully if parking lot is full. Currently, if a parking lot is full, bridge setup returns -1, causing dialplan execution to terminate without TryExec. However, such failures should be handled more gracefully, the same way they are on other paths, as indicated by the module's author, here: http://lists.digium.com/pipermail/asterisk-dev/2018-December/077144.html Now, callers will hear the parking failure announcement, and dialplan will continue, which is consistent with existing failure modes. Resolves: #624	1 year ago
Sean Bright	78471aa1a5	res_config_mysql.c: Support hostnames up to 255 bytes. Fixes #654	1 year ago
Sean Bright	2aa7f7a4dc	res_pjsip: Fix alembic downgrade for boolean columns. When downgrading, ensure that we don't touch columns that didn't actually change during upgrade.	1 year ago
Stanislav Abramenkov	929bc2693c	Upgrade bundled pjproject to 2.14.1 Fixes: asterisk#648 UserNote: Bundled pjproject has been upgraded to 2.14.1. For more information visit pjproject Github page: https://github.com/pjsip/pjproject/releases/tag/2.14.1	1 year ago
Sean Bright	957cde97fd	alembic: Quote new MySQL keyword 'qualify.' Fixes #651	1 year ago
Maximilian Fridrich	f626af8e13	res_pjsip_session: Reset pending_media_state->read_callbacks In handle_negotiated_sdp the pending_media_state->read_callbacks must be reset before they are added in the SDP handlers in handle_negotiated_sdp_session_media. Otherwise, old callbacks for removed streams and file descriptors could be added to the channel and Asterisk would poll on non-existing file descriptors. Resolves: #611	1 year ago
Sean Bright	3c1a2daecd	res_monitor.c: Don't emit a warning about 'X' being unrecognized. Code was added in `030f7d41` to warn if an unrecognized option was passed to an application, but code in Monitor was taking advantage of the fact that the application would silently accept an invalid option. We now recognize the invalid option but we don't do anything if it's set. Fixes #639	1 year ago
George Joseph	61b5298cd3	res_pjsip_stir_shaken.c: Add checks for missing parameters * Added checks for missing session, session->channel and rdata in stir_shaken_incoming_request. * Added checks for missing session, session->channel and tdata in stir_shaken_outgoing_request. Resolves: #645	1 year ago

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33960 Commits (18) All Branches Search

33960 Commits (18)

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