asterisk

Commit Graph

Author	SHA1	Message	Date
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
Martin Nystroem	dd91f09481	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	2 years ago
Sean Bright	1944c9d72e	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.	2 years ago
jonatascalebe	d4b79cb466	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()	2 years ago
Naveen Albert	4d45b84dee	menuselect: Minor cosmetic fixes. Improve some of the formatting from `dd3f17c699` (#521).	2 years ago
Naveen Albert	310c8a7c68	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.	2 years ago
Holger Hans Peter Freyther	d45c8e165f	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	2 years ago
Naveen Albert	f4fba80708	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.	2 years ago
Naveen Albert	7f51313725	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.	2 years ago
Naveen Albert	e1dfa20797	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	2 years ago
Sean Bright	e33bab1bfd	res_config_mysql.c: Support hostnames up to 255 bytes. Fixes #654	2 years ago
Sean Bright	ce8b564509	res_pjsip: Fix alembic downgrade for boolean columns. When downgrading, ensure that we don't touch columns that didn't actually change during upgrade.	2 years ago
Stanislav Abramenkov	72310b5575	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	2 years ago
Sean Bright	17cf90cc8b	alembic: Quote new MySQL keyword 'qualify.' Fixes #651	2 years ago
Maximilian Fridrich	c5a6d8a6db	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	2 years ago
George Joseph	51790abba7	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	2 years ago
George Joseph	a02a929d27	.github: Add PAT to PRSubmitActions/Add Reviewers	2 years ago
Naveen Albert	bdf9327a33	app_dial: Add dial time for progress/ringing. Add a timeout option to control the amount of time to wait if no early media is received before giving up. This allows aborting early if the destination is not being responsive. Resolves: #588 UserNote: The timeout argument to Dial now allows specifying the maximum amount of time to dial if early media is not received.	2 years ago
Naveen Albert	2ac9c8fb5c	app_voicemail: Properly reinitialize config after unit tests. Most app_voicemail unit tests were not properly cleaning up after themselves after running. This led to test mailboxes lingering around in the system. It also meant that if any unit tests in app_voicemail that create mailboxes were executed and the module was not unloaded/loaded again prior to running the test_voicemail_vm_info unit test, Asterisk would segfault due to an attempt to copy a NULL string. The load_config test did actually have logic to reinitialize the config after the test. However, this did not work in practice since load_config() would not reload the config since voicemail.conf had not changed during the test; thus, additional logic has been added to ensure that voicemail.conf is truly reloaded, after any unit tests which modify the users list. This prevents the SEGV due to invalid mailboxes lingering around, and also ensures that the system state is restored to what it was prior to the tests running. Resolves: #629	2 years ago
Shaaah	6f99268f79	app_queue.c : fix "queue add member" usage string Fixing bracket placement in the "queue add member" cli usage string.	2 years ago
Naveen Albert	190b6eafb3	app_voicemail: Allow preventing mark messages as urgent. This adds an option to allow preventing callers from leaving messages marked as 'urgent'. Resolves: #619 UserNote: The leaveurgent mailbox option can now be used to control whether callers may leave messages marked as 'Urgent'.	2 years ago
Sean Bright	1a94502551	res_pjsip: Use consistent type for boolean columns. This migrates the relevant schema objects from the `('yes', 'no')` definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')` one. Fixes #617	2 years ago
George Joseph	94176ecb88	.github: Remove timeout-minutes from gatetests	2 years ago
George Joseph	1b94c90524	attestation_config.c: Use ast_free instead of ast_std_free In as_check_common_config, we were calling ast_std_free on raw_key but raw_key was allocated with ast_malloc so it should be freed with ast_free. Resolves: #636	2 years ago
George Joseph	215424aa76	Makefile: Add stir_shaken/cache to directories created on install The default location for the stir_shaken cache is /var/lib/asterisk/keys/stir_shaken/cache but we were only creating /var/lib/asterisk/keys/stir_shaken on istall. We now create the cache sub-directory. Resolves: #634	2 years ago
George Joseph	d58ca20247	.github: Pass only single GATETEST_COMMAND to AsteriskGateComposite	2 years ago
George Joseph	2e0d837e01	Stir/Shaken Refactor Why do we need a refactor? The original stir/shaken implementation was started over 3 years ago when little was understood about practical implementation. The result was an implementation that wouldn't actually interoperate with any other stir-shaken implementations. There were also a number of stir-shaken features and RFC requirements that were never implemented such as TNAuthList certificate validation, sending Reason headers in SIP responses when verification failed but we wished to continue the call, and the ability to send Media Key(mky) grants in the Identity header when the call involved DTLS. Finally, there were some performance concerns around outgoing calls and selection of the correct certificate and private key. The configuration was keyed by an arbitrary name which meant that for every outgoing call, we had to scan the entire list of configured TNs to find the correct cert to use. With only a few TNs configured, this wasn't an issue but if you have a thousand, it could be. What's changed? * Configuration objects have been refactored to be clearer about their uses and to fix issues. * The "general" object was renamed to "verification" since it contains parameters specific to the incoming verification process. It also never handled ca_path and crl_path correctly. * A new "attestation" object was added that controls the outgoing attestation process. It sets default certificates, keys, etc. * The "certificate" object was renamed to "tn" and had it's key change to telephone number since outgoing call attestation needs to look up certificates by telephone number. * The "profile" object had more parameters added to it that can override default parameters specified in the "attestation" and "verification" objects. * The "store" object was removed altogther as it was never implemented. * We now use libjwt to create outgoing Identity headers and to parse and validate signatures on incoming Identiy headers. Our previous custom implementation was much of the source of the interoperability issues. * General code cleanup and refactor. * Moved things to better places. * Separated some of the complex functions to smaller ones. * Using context objects rather than passing tons of parameters in function calls. * Removed some complexity and unneeded encapsuation from the config objects. Resolves: #351 Resolves: #46 UserNote: Asterisk's stir-shaken feature has been refactored to correct interoperability, RFC compliance, and performance issues. See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information. UpgradeNote: The stir-shaken refactor is a breaking change but since it's not working now we don't think it matters. The stir_shaken.conf file has changed significantly which means that existing ones WILL need to be changed. The stir_shaken.conf.sample file in configs/samples/ has quite a bit more information. This is also an ABI breaking change since some of the existing objects needed to be changed or removed, and new ones added. Additionally, if res_stir_shaken is enabled in menuselect, you'll need to either have the development package for libjwt v1.15.3 installed or use the --with-libjwt-bundled option with ./configure.	2 years ago
Sebastian Jennen	3c72bc8a7b	translate.c: implement new direct comp table mode The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio. This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing). - add new table mode - hide the 999999 comp values, as these only indicate an issue with transcoding - hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding) Resolves: #601	2 years ago
Shyju Kanaprath	07055dce94	README.md: Removed outdated link Removed outdated link http://www.quicknet.net from README.md cherry-pick-to: 18 cherry-pick-to: 20 cherry-pick-to: 21	2 years ago
Sean Bright	31ab82840b	strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string. If a dynamic string is created with an initial length of 0, `ast_str_buffer(…)` will return an invalid pointer. This was a secondary discovery when fixing #65.	2 years ago
George Joseph	2d1a4bab25	.github: Add force_cherry_pick option to Releaser	2 years ago
George Joseph	4077b4b340	.github: Remove start_version from Releaser	2 years ago
romryz	a7a03bc294	res_rtp_asterisk.c: Correct coefficient in MOS calculation. Media Experience Score relies on incorrect pseudo_mos variable calculation. According to forming an opinion section of the documentation, calculation relies on ITU-T G.107 standard: https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion ITU-T G.107 Annex B suggests to calculate MOS with a coefficient "seven times ten to the power of negative six", 7 * 10^(-6). which would mean 6 digits after the decimal point. Current implementation has 7 digits after the decimal point, which downrates the calls. Fixes: #597	2 years ago
Naveen Albert	6ddcdfce1f	dsp.c: Fix and improve potentially inaccurate log message. If ast_dsp_process is called with a codec besides slin, ulaw, or alaw, a warning is logged that in-band DTMF is not supported, but this message is not always appropriate or correct, because ast_dsp_process is much more generic than just DTMF detection. This logs a more generic message in those cases, and also improves codec-mismatch logging throughout dsp.c by ensuring incompatible codecs are printed out. Resolves: #595	2 years ago
George Joseph	adcfbcd50e	pjsip show channelstats: Prevent possible segfault when faxing Under rare circumstances, it's possible for the original audio session in the active_media_state default_session to be corrupted instead of removed when switching to the t38/image media session during fax negotiation. This can cause a segfault when a "pjsip show channelstats" attempts to print that audio media session's rtp statistics. In these cases, the active_media_state topology is correctly showing only a single t38/image stream so we now check that there's an audio stream in the topology before attempting to use the audio media session to get the rtp statistics. Resolves: #592	2 years ago
George Joseph	a5ae546b88	Reduce startup/shutdown verbose logging When started with a verbose level of 3, asterisk can emit over 1500 verbose message that serve no real purpose other than to fill up logs. When asterisk shuts down, it emits another 1100 that are of even less use. Since the testsuite runs asterisk with a verbose level of 3, and asterisk starts and stops for every one of the 700+ tests, the number of log messages is staggering. Besides taking up resources, it also makes it hard to debug failing tests. This commit changes the log level for those verbose messages to 5 instead of 3 which reduces the number of log messages to only a handful. Of course, NOTICE, WARNING and ERROR message are unaffected. There's also one other minor change... ast_context_remove_extension_callerid2() logs a DEBUG message instead of an ERROR if the extension you're deleting doesn't exist. The pjsip_config_wizard calls that function to clean up the config and has been triggering that annoying error message for years. Resolves: #582	2 years ago
Naveen Albert	d0d09ef010	configure: Rerun bootstrap on modern platform. The last time configure was run, it was run on a system that did not enable -std=gnu11 by default, which meant that the restrict qualifier would not be recognized on certain platforms. This regenerates the configure files from running bootstrap.sh, so that these should be recognized on all supported platforms. Resolves: #586	2 years ago
Ben Ford	6efa51f512	Upgrade bundled pjproject to 2.14. Fixes: #406 UserNote: Bundled pjproject has been upgraded to 2.14. For more information on what all is included in this change, check out the pjproject Github page: https://github.com/pjsip/pjproject/releases	2 years ago
Flole998	c7fc6ae362	res_pjsip_outbound_registration.c: Add User-Agent header override This introduces a setting for outbound registrations to override the global User-Agent header setting. Resolves: #515 UserNote: PJSIP outbound registrations now support a per-registration User-Agent header	2 years ago
cmaj	c863e0d77d	app_speech_utils.c: Allow partial speech results. Adds 'p' option to SpeechBackground() application. With this option, when the app timeout is reached, whatever the backend speech engine collected will be returned as if it were the final, full result. (This works for engines that make partial results.) Resolves: #572 UserNote: The SpeechBackground dialplan application now supports a 'p' option that will return partial results from speech engines that provide them when a timeout occurs.	2 years ago
Joshua C. Colp	8ce69eda14	utils: Make behavior of ast_strsep* match strsep. Given the scenario of passing an empty string to the ast_strsep functions the functions would return NULL instead of an empty string. This is counter to how strsep itself works. This change alters the behavior of the functions to match that of strsep. Fixes: #565	2 years ago
Mike Bradeen	69fe814813	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

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