asterisk

Commit Graph

Author	SHA1	Message	Date
mkmer	68d9a69be3	audiohook.c: Add ability to adjust volume with float Add the capability to audiohook for float type volume adjustments. This allows for adjustments to volume smaller than 6dB. With INT adjustments, the first step is 2 which converts to ~6dB (or 1/2 volume / double volume depending on adjustment sign). 3dB is a typical adjustment level which can now be accommodated with an adjustment value of 1.41. This is accomplished by the following: Convert internal variables to type float. Always use ast_frame_adjust_volume_float() for adjustments. Cast int to float in original functions ast_audiohook_volume_set(), and ast_volume_adjust(). Cast float to int in ast_audiohook_volume_get() Add functions ast_audiohook_volume_get_float, ast_audiohook_volume_set_float, and ast_audiohook_volume_adjust_float. This update maintains 100% backward compatibility. Resolves: #1171 (cherry picked from commit `ecd1a727e8`)	6 months ago
Florent CHAUVEAU	a15e4043ce	audiosocket: added support for DTMF frames Updated the AudioSocket protocol to allow sending DTMF frames. AST_FRAME_DTMF frames are now forwarded to the server, in addition to AST_FRAME_AUDIO frames. A new payload type AST_AUDIOSOCKET_KIND_DTMF with value 0x03 was added to the protocol. The payload is a 1-byte ascii representing the DTMF digit (0-9,,#...). UserNote: The AudioSocket protocol now forwards DTMF frames with payload type 0x03. The payload is a 1-byte ascii representing the DTMF digit (0-9,,#...). (cherry picked from commit `c2444b188a`)	6 months ago
Norm Harrison	352a72c9b5	asterisk/channel.h: fix documentation for 'ast_waitfor_nandfds()' Co-authored-by: Florent CHAUVEAU <florentch@pm.me> (cherry picked from commit `6c6195ce4d`)	6 months ago
Norm Harrison	d045f290b7	audiosocket: fix timeout, fix dialplan app exit, server address in logs - Correct wait timeout logic in the dialplan application. - Include server address in log messages for better traceability. - Allow dialplan app to exit gracefully on hangup messages and socket closure. - Optimize I/O by reducing redundant read()/write() operations. Co-authored-by: Florent CHAUVEAU <florentch@pm.me> (cherry picked from commit `a6ef41e4ca`)	6 months ago
Mark Murawski	d18b6bb9b3	chan_pjsip: Add the same details as PJSIPShowContacts to the CLI via 'pjsip show contact' CLI 'pjsip show contact' does not show enough information. One must telnet to AMI or write a script to ask Asterisk for example what the User-Agent is on a Contact This feature adds the same details as PJSIPShowContacts to the CLI Resolves: #643 (cherry picked from commit `fa69682c28`)	6 months ago
Zhai Liangliang	419423f68c	Update config.guess and config.sub (cherry picked from commit `ea37d6a6e1`)	6 months ago
Alexei Gradinari	04ed031b62	chan_pjsip: set correct Endpoint Device State on multiple channels 1. When one channel is placed on hold, the device state is set to ONHOLD without checking other channels states. In case of AST_CONTROL_HOLD set the device state as AST_DEVICE_UNKNOWN to calculate aggregate device state of all active channels. 2. The current implementation incorrectly classifies channels in use. The only channels that has the states: UP, RING and BUSY are considered as "in use". A channel should be considered "in use" if its state is anything other than DOWN or RESERVED. 3. Currently, if the number of channels "in use" is greater than device_state_busy_at, the system does not set the state to BUSY. Instead, it incorrectly assigns an aggregate device state. The endpoint device state should be BUSY if the number of channels "in use" is greater than or equal to device_state_busy_at. Fixes: #1181 (cherry picked from commit `b82d9b0d5b`)	6 months ago
Allan Nathanson	bdc925c8cc	file.c: missing "custom" sound files should not generate warning logs With `sounds_search_custom_dir = yes` we first look to see if a sound file is present in the "custom" sound directory before looking in the standard sound directories. We should not be issuing a WARNING log message if a sound cannot be found in the "custom" directory. Resolves: https://github.com/asterisk/asterisk/issues/1170 (cherry picked from commit `e2e36748f5`)	6 months ago
Asterisk Development Team	b7f7d66fca	Update for 22.3.0	7 months ago
Asterisk Development Team	6fed735157	Update for 22.3.0-rc1	7 months ago
Ben Ford	10ef135e5a	documentation: Update Gosub, Goto, and add new documentationtype. Gosub and Goto were not displaying their syntax correctly on the docs site. This change adds a new way to specify an optional context, an optional extension, and a required priority that the xml stylesheet can parse without having to know which optional parameters come in which order. In Asterisk, it looks like this: parameter name="context" documentationtype="dialplan_context" parameter name="extension" documentationtype="dialplan_extension" parameter name="priority" documentationtype="dialplan_priority" required="true" The stylesheet will ignore the context and extension parameters, but for priority, it will automatically inject the following: [[context,]extension,]priority This is the correct oder for applications such as Gosub and Goto. (cherry picked from commit `091171c0eb`)	7 months ago
Sean Bright	d9f1b1598f	res_config_curl.c: Remove unnecessary warnings. Resolves: #1164 (cherry picked from commit `0900f83820`)	7 months ago
George Joseph	dce0ae89a4	README.md: Updates and Fixes * Outdated information has been removed. * New links added. * Placeholder added for link to change logs. Going forward, the release process will create HTML versions of the README and change log and will update the link in the README to the current change log for the branch... * In the development branches, the link will always point to the current release on GitHub. * In the "releases/" branches and the tarballs, the link will point to the ChangeLogs/ChangeLog-<version>.html file in the source directory. On the downloads website, the link will point to the ChangeLog-<version>.html file in the same directory. Resolves: #1131 (cherry picked from commit `dc5c545175`)	7 months ago
Sean Bright	ff8ec30efb	res_rtp_asterisk.c: Don't truncate spec-compliant `ice-ufrag` or `ice-pwd`. RFC 8839[1] indicates that the `ice-ufrag` and `ice-pwd` attributes can be up to 256 bytes long. While we don't generate values of that size, we should be able to accomodate them without truncating. 1. https://www.rfc-editor.org/rfc/rfc8839#name-ice-ufrag-and-ice-pwd-attri (cherry picked from commit `25aff1a887`)	7 months ago
Joshua Elson	4a1129b893	fix: Correct default flag for tcp_keepalive_enable option Resolves an issue where the tcp_keepalive_enable option was not properly enabled in the sample configuration due to an incorrect default flag setting. Fixes: #1149 (cherry picked from commit `c4232f2940`)	7 months ago
Sean Bright	8e900950b6	docs: AMI documentation fixes. Most of this patch is adding missing PJSIP-related event documentation, but the one functional change was adding a sorcery to-string handler for endpoint's `redirect_method` which was not showing up in the AMI event details or `pjsip show endpoint <endpoint>` output. The rest of the changes are summarized below: * app_agent_pool.c: Typo fix Epoche -> Epoch. * stasis_bridges.c: Add missing AttendedTransfer properties. * stasis_channels.c: Add missing AgentLogoff properties. * pjsip_manager.xml: - Add missing AorList properties. - Add missing AorDetail properties. - Add missing ContactList properties. - Add missing ContactStatusDetail properties. - Add missing EventDetail properties. - Add missing AuthList properties. - Add missing AuthDetail properties. - Add missing TransportDetail properties. - Add missing EndpointList properties. - Add missing IdentifyDetail properties. * res_pjsip_registrar.c: Add missing InboundRegistrationDetail documentation. * res_pjsip_pubsub.c: - Add missing ResourceListDetail documentation. - Add missing InboundSubscriptionDetail documentation. - Add missing OutboundSubscriptionDetail documentation. * res_pjsip_outbound_registration.c: Add missing OutboundRegistrationDetail documentation. (cherry picked from commit `f042fb2153`)	7 months ago
Allan Nathanson	2d75ac2c0e	config.c: #include of non-existent file should not crash Corrects a segmentation fault when a configuration file has a #include statement that referenced a file that does not exist. Resolves: https://github.com/asterisk/asterisk/issues/1139 (cherry picked from commit `ffb0969a98`)	7 months ago
George Joseph	d184ae428d	manager.c: Check for restricted file in action_createconfig. The `CreateConfig` manager action now ensures that a config file can only be created in the AST_CONFIG_DIR unless `live_dangerously` is set. Resolves: #1122 (cherry picked from commit `d4ece751c5`)	7 months ago
George Joseph	77c6bf844f	swagger_model.py: Fix invalid escape sequence in get_list_parameter_type(). Recent python versions complain when backslashes in strings create invalid escape sequences. This causes issues for strings used as regex patterns like `'^List\[(.)\]$'` where you want the regex parser to treat `[` and `]` as literals. Double-backslashing is one way to fix it but simply converting the string to a raw string `re.match(r'^List\[(.)\]$', text)` is easier and less error prone. (cherry picked from commit `c30ffc08a7`)	7 months ago
Maximilian Fridrich	99ef285fff	Revert "res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big" This reverts commit `f30ad96b3f`. The original change was not RFC compliant and caused issues because it set the RTP marker bit in cases when it shouldn't be set. See the linked issue #1135 for a detailed explanation. Fixes: #1135. (cherry picked from commit `44ca269adb`)	7 months ago
Sean Bright	f5e87a6da3	res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer. Found while reviewing #1128 (cherry picked from commit `47e2e59230`)	7 months ago
Luz Paz	540e28ea8d	docs: Fix typos in cdr/ Found via codespell (cherry picked from commit `f221ef9777`)	7 months ago
Luz Paz	350a80a2aa	docs: Fix various typos in channels/ Found via `codespell -q 3 -S "./CREDITS,*.po" -L abd,asent,atleast,cachable,childrens,contentn,crypted,dne,durationm,enew,exten,inout,leapyear,mye,nd,oclock,offsetp,ot,parm,parms,preceeding,pris,ptd,requestor,re-use,re-used,re-uses,ser,siz,slanguage,slin,thirdparty,varn,varns,ues` (cherry picked from commit `32cdce18c2`)	7 months ago
Luz Paz	3692e33898	docs: Fix various typos in main/ Found via `codespell -q 3 -S "./CREDITS" -L abd,asent,atleast,childrens,contentn,crypted,dne,durationm,exten,inout,leapyear,nd,oclock,offsetp,ot,parm,parms,requestor,ser,slanguage,slin,thirdparty,varn,varns,ues` (cherry picked from commit `b96640916d`)	7 months ago
George Joseph	de3d6c03e3	bridging: Fix multiple bridging issues causing SEGVs and FRACKs. Issues: * The bridging core allowed multiple bridges to be created with the same unique bridgeId at the same time. Only the last bridge created with the duplicate name was actually saved to the core bridges container. * The bridging core was creating a stasis topic for the bridge and saving it in the bridge->topic field but not increasing its reference count. In the case where two bridges were created with the same uniqueid (which is also the topic name), the second bridge would get the _existing_ topic the first bridge created. When the first bridge was destroyed, it would take the topic with it so when the second bridge attempted to publish a message to it it either FRACKed or SEGVd. * The bridge destructor, which also destroys the bridge topic, is run from the bridge manager thread not the caller's thread. This makes it possible for an ARI developer to create a new one with the same uniqueid believing the old one was destroyed when, in fact, the old one's destructor hadn't completed. This could cause the new bridge to get the old one's topic just before the topic was destroyed. When the new bridge attempted to publish a message on that topic, asterisk could either FRACK or SEGV. * The ARI bridges resource also allowed multiple bridges to be created with the same uniqueid but it kept the duplicate bridges in its app_bridges container. This created a situation where if you added two bridges with the same "bridge1" uniqueid, all operations on "bridge1" were performed on the first bridge created and the second was basically orphaned. If you attempted to delete what you thought was the second bridge, you actually deleted the first one created. Changes: * A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if a topic already exists for a bridge. * `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check if a bridge with the requested uniqueid already exists and will fail if it does. * `bridge_register()` in bridges.c now checks the core bridges container to make sure a bridge doesn't already exist with the requested uniqueid. Although most callers of `bridge_register()` will have already called `bridge_base_init()`, which will now fail on duplicate bridges, there is no guarantee of this so we must check again. * The core bridges container allocation was changed to reject duplicate uniqueids instead of silently replacing an existing one. This is a "belt and suspenders" check. * A global mutex was added to bridge.c to prevent concurrent calls to `bridge_base_init()` and `bridge_register()`. * Even though you can no longer create multiple bridges with the same uniqueid at the same time, it's still possible that the bridge topic might be destroyed while a second bridge with the same uniqueid was trying to use it. To address this, the bridging core now increments the reference count on bridge->topic when a bridge is created and decrements it when the bridge is destroyed. * `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges container to make sure a bridge with the requested uniqueid doesn't already exist. This may seem like overkill but there are so many entrypoints to bridge creation that we need to be safe and catch issues as soon in the process as possible. * The stasis app_bridges container allocation was changed to reject duplicate uniqueids instead of adding them. This is a "belt and suspenders" check. * The `bridge show all` CLI command now shows the bridge name as well as the bridge id. * Response code 409 "Conflict" was added as a possible response from the ARI bridge create resources to signal that a bridge with the requested uniqueid already exists. * Additional debugging was added to multiple bridging and stasis files. Resolves: #211 (cherry picked from commit `169b9b9a4f`)	7 months ago
George Joseph	fc1de1f290	.github: Change concurrency group ids so they're unique. GitHub strikes again. Apparently the github.ref context variable only contains the PR number if the workflow is triggered by "pull_request" so since we just changed the trigger to "pull_request_target" the variable no longer contains the PR number and is therefore not unique and can't be used as a concurrency group id. We now use `github.triggering_actor-github.head_ref`. (cherry picked from commit `313608837e`)	7 months ago
Mike Bradeen	ee09b289ab	bridge_channel: don't set cause code on channel during bridge delete if already set Due to a potential race condition via ARI when hanging up a channel hangup with cause while also deleting a bridge containing that channel, the bridge delete can over-write the hangup cause code resulting in Normal Call Clearing instead of the set value. With this change, bridge deletion will only set the hangup code if it hasn't been previously set. Resolves: #1124 (cherry picked from commit `7e694417ad`)	7 months ago
George Joseph	5ddfbd5eab	.github: Refactor Releaser to use reusable workflow (cherry picked from commit `aa9d6fb9e3`)	7 months ago
George Joseph	d6ce7bb395	.github: Change branch of reusable workflows to main. (cherry picked from commit `07f37f51fe`)	7 months ago
George Joseph	02bdc2dc37	.github: Refactor to use pull_request_target trigger. After careful review, we believe we can now use the "pull_request_target" workflow trigger instead of "pull_request" which required a separate privliged workflow to add labels and comments to PRs when they are submitted or updated. This allows us to greatly streamline our workflows and remove unneeded ones. * The OnPRChanged workflow was... * Renamed to OnPRCheck * Changed to trigger on pull_request_target and the "recheckpr" label. * Changed to simply call reusable workflows in asterisk-ci-actions. * Changed to use better concurrency groups. * The OnPRCPCheck and OnPRMergeApproved workflows were also... * Changed to simply call reusable workflows in asterisk-ci-actions. * Changed to use better concurrency groups. * The NightlyTest and CreateDocs were also tweaked (cherry picked from commit `ebb33475bc`)	7 months ago
George Joseph	26ea320f3c	res_config_pgsql: Fix regression that removed dbname config. A recent commit accidentally removed the code that sets dbname. This commit adds it back in. Resolves: #1119 (cherry picked from commit `aef37c2a65`)	7 months ago
George Joseph	3ec2a680f2	res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan. The verification check for missing or anonymous callerid was happening before the endpoint's profile was retrieved which meant that the failure_action parameter wasn't available. Therefore, if verification was enabled and there was no callerid or it was "anonymous", the call was immediately terminated instead of giving the dialplan the ability to decide what to do with the call. * The callerid check now happens after the verification context is created and the endpoint's stir_shaken_profile is available. * The check now processes the callerid failure just as it does for other verification failures and respects the failure_action parameter. If set to "continue" or "continue_return_reason", `STIR_SHAKEN(0,verify_result)` in the dialplan will return "invalid_or_no_callerid". * If the endpoint's failure_action is "reject_request", the call will be rejected with `433 "Anonymity Disallowed"`. * If the endpoint's failure_action is "continue_return_reason", the call will continue but a `Reason: STIR; cause=433; text="Anonymity Disallowed"` header will be added to the next provisional or final response. Resolves: #1112 (cherry picked from commit `85fa81ad45`)	7 months ago
George Joseph	619cb75688	resource_channels.c: Fix memory leak in ast_ari_channels_external_media. Between ast_ari_channels_external_media(), external_media_rtp_udp(), and external_media_audiosocket_tcp(), the `variables` structure being passed around wasn't being cleaned up properly when there was a failure. * In ast_ari_channels_external_media(), the `variables` structure is now defined with RAII_VAR to ensure it always gets cleaned up. * The ast_variables_destroy() call was removed from external_media_rtp_udp(). * The ast_variables_destroy() call was removed from external_media_audiosocket_tcp(), its `endpoint` allocation was changed to to use ast_asprintf() as external_media_rtp_udp() does, and it now returns an error on failure. * ast_ari_channels_external_media() now checks the new return code from external_media_audiosocket_tcp() and sets the appropriate error response. Resolves: #1109 (cherry picked from commit `1a1eb88211`)	7 months ago
Holger Hans Peter Freyther	fe5ff3021e	ari/pjsip: Make it possible to control transfers through ARI Introduce a ChannelTransfer event and the ability to notify progress to ARI. Implement emitting this event from the PJSIP channel instead of handling the transfer in Asterisk when configured. Introduce a dialplan function to the PJSIP channel to switch between the "core" and "ari-only" behavior. UserNote: Call transfers on the PJSIP channel can now be controlled by ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) dialplan function. (cherry picked from commit `a0d0c47d06`)	7 months ago
George Joseph	ba1626df89	.github: Remove concurrency check in on-labelled workflows. Apparently you can't use `${{ github.event.number }}` in a concurrency block in a job that calls a reusable workflow. :( (cherry picked from commit `d53f96b6af`)	7 months ago
Sean Bright	8f97da052d	channel.c: Remove dead AST_GENERATOR_FD code. Nothing ever sets the `AST_GENERATOR_FD`, so this block of code will never execute. It also is the only place where the `generate` callback is called with the channel lock held which made it difficult to reason about the thread safety of `ast_generator`s. In passing, also note that `AST_AGENT_FD` isn't used either. (cherry picked from commit `46017ea15b`)	7 months ago
George Joseph	86ff4b8a5b	.github: Move PRChanged,PRChangedPriv,PRCPCheck,PRReCheck,PRMerge logic. Moved to asterisk-ci-actions reusable workflows. (cherry picked from commit `595739a21c`)	7 months ago
George Joseph	2524043b61	.github: OnPRCherryPickTest,OnPRStateChanged,OnPRRecheck: Add job summaries. ...and refactor environment variables. (cherry picked from commit `9b1616e501`)	7 months ago
George Joseph	26ad51c2ff	.github: Clean up CreateDocs (cherry picked from commit `ad0f268c5b`)	7 months ago
George Joseph	ecb3c3d8e5	func_strings.c: Prevent SEGV in HASH single-argument mode. When in single-argument mode (very rarely used), a malformation of a column name (also very rare) could cause a NULL to be returned when retrieving the channel variable for that column. Passing that to strncat causes a SEGV. We now check for the NULL and print a warning message. Resolves: #1101 (cherry picked from commit `6574f6532d`)	7 months ago
George Joseph	7863ff70bd	docs: Add version information to AGI command XML elements. This process was a bit different than the others because everything is in the same file, there's an array that contains the command names and their handler functions, and the last command was created over 15 years ago. * Dump a `git blame` of res/res_agi.c from BEFORE the handle_* prototypes were changed. * Create a command <> handler function xref by parsing the the agi_command array. * For each entry, grep the function definition line "static int handle_" from the git blame output and capture the commit. This will be the commit the command was created in. Do a `git tag --contains <commit> \| sort -V \| head -1` to get the tag the function was created in. * Add a single since/version element to the command XML. Multiple versions aren't supported here because the branching and tagging scheme changed several times in the 2000's. (cherry picked from commit `8dd5c23494`)	7 months ago
Jeremy Lainé	47d3cca627	docs: Fix minor typo in MixMonitor AMI action The `Options` argument was erroneously documented as lowercase `options`. (cherry picked from commit `85538b6d71`)	7 months ago
Naveen Albert	c0ed22368a	utils: Disable old style definition warnings for libdb. Newer versions of gcc now warn about old style definitions, such as those in libdb, which causes compilation failure with DEVMODE enabled. Ignore these warnings for libdb. Resolves: #1085 (cherry picked from commit `5b2f36ffae`)	7 months ago
fabriziopicconi	b31d457d0a	rtp.conf.sample: Correct stunaddr example. (cherry picked from commit `4d9a73af71`)	7 months ago
George Joseph	8d386d7c8b	docs: Add version information to ARI resources and methods. * Dump a git blame of each file in rest-api/api-docs. * Get the commit for each "resourcePath" and "httpMethod" entry. * Find the tags for each commit (same as other processes). * Insert a "since" array after each "resourcePath" and "httpMethod" entry. (cherry picked from commit `bbc0a609fc`)	7 months ago
Sean Bright	8a3a2958c1	docs: Indent <since> tags. Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0). (cherry picked from commit `67e89b3e77`)	7 months ago
Asterisk Development Team	a0db68c190	Update for 22.2.0	9 months ago
Asterisk Development Team	eb782ec341	Update for 22.2.0-rc2	9 months ago
George Joseph	5ac0874240	res_pjsip_authenticator_digest: Make correct error messages appear again. When an incoming request can't be matched to an endpoint, the "artificial" auth object is used to create a challenge to return in a 401 response and we emit a "No matching endpoint found" log message. If the client then responds with an Authorization header but the request still can't be matched to an endpoint, the verification will fail and, as before, we'll create a challenge to return in a 401 response and we emit a "No matching endpoint found" log message. HOWEVER, because there WAS an Authorization header and it failed verification, we should have also been emitting a "Failed to authenticate" log message but weren't because there was a check that short-circuited that it if the artificial auth was used. Since many admins use the "Failed to authenticate" message with log parsers like fail2ban, those attempts were not being recognized as suspicious. Changes: * digest_check_auth() now always emits the "Failed to authenticate" log message if verification of an Authorization header failed even if the artificial auth was used. * The verification logic was refactored to be clearer about the handling of the return codes from verify(). * Comments were added clarify what return codes digest_check_auth() should return to the distributor and the implications of changing them. Resolves: #1095	9 months ago
George Joseph	c3cf4bd223	alembic: Database updates required. This commit doesn't actually change anything. It just adds the following upgrade notes that were omitted from the original commits. Resolves: #1097 UpgradeNote: Two commits in this release... 'Add SHA-256 and SHA-512-256 as authentication digest algorithms' 'res_pjsip: Add new AOR option "qualify_2xx_only"' ...have modified alembic scripts for the following database tables: ps_aors, ps_contacts, ps_auths, ps_globals. If you don't use the scripts to update your database, reads from those tables will succeeed but inserts into the ps_contacts table by res_pjsip_registrar will fail.	9 months ago

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