5562 Commits (2bb607f7b74332a83aea6e5200c7569ea707d16b)

Author	SHA1	Message	Date
Naveen Albert	2bb607f7b7	res_pjsip_caller_id: Also parse URI parameters for ANI2. ... If the isup-oli was sent as a URI parameter, rather than a header parameter, it was not being parsed. Make sure we parse both if needed so the ANI2 is set regardless of which type of parameter the isup-oli is sent as. Resolves: #1220	9 months ago
Mike Bradeen	4dc3ca4c9a	stasis/control.c: Set Hangup Cause to No Answer on Dial timeout ... Other Dial operations (dial, app_dial) use Q.850 cause 19 when a dial timeout occurs, but the Dial command via ARI did not set an explicit reason. This resulted in a CANCEL with Normal Call Clearing and corresponding ChannelDestroyed. This change sets the hangup cause to AST_CAUSE_NO_ANSWER to be consistent with the other operations. Fixes: #963 UserNote: A Dial timeout on POST /channels/{channelId}/dial will now result in a CANCEL and ChannelDestroyed with cause 19 / User alerting, no answer. Previously no explicit cause was set, resulting in a cause of 16 / Normal Call Clearing.	9 months ago
Albrecht Oster	c251afadb9	res_pjproject: Fix DTLS client check failing on some platforms ... Certain platforms (mainly BSD derivatives) have an additional length field in `sockaddr_in6` and `sockaddr_in`. `ast_sockaddr_from_pj_sockaddr()` does not take this field into account when copying over values from the `pj_sockaddr` into the `ast_sockaddr`. The resulting `ast_sockaddr` will have an uninitialized value for `sin6_len`/`sin_len` while the other `ast_sockaddr` (not converted from a `pj_sockaddr`) to check against in `ast_sockaddr_pj_sockaddr_cmp()` has the correct length value set. This has the effect that `ast_sockaddr_cmp()` will always indicate an address mismatch, because it does a bitwise comparison, and all DTLS packets are dropped even if addresses and ports match. `ast_sockaddr_from_pj_sockaddr()` now checks whether the length fields are available on the current platform and sets the values accordingly. Resolves: #505	9 months ago
George Joseph	f8bc3ddeb9	Prequisites for ARI Outbound Websockets ... stasis: * Added stasis_app_is_registered(). * Added stasis_app_control_mark_failed(). * Added stasis_app_control_is_failed(). * Fixed res_stasis_device_state so unsubscribe all works properly. * Modified stasis_app_unregister() to unsubscribe from all event sources. * Modified stasis_app_exec to return -1 if stasis_app_control_is_failed() returns true. http: * Added ast_http_create_basic_auth_header(). md5: * Added define for MD5_DIGEST_LENGTH. tcptls: * Added flag to ast_tcptls_session_args to suppress connection log messages to give callers more control over logging. http_websocket: * Add flag to ast_websocket_client_options to suppress connection log messages to give callers more control over logging. * Added username and password to ast_websocket_client_options to support outbound basic authentication. * Added ast_websocket_result_to_str().	9 months ago
George Joseph	62e73f9bd8	ari_websockets: Fix frack if ARI config fails to load. ... ari_ws_session_registry_dtor() wasn't checking that the container was valid before running ao2_callback on it to shutdown registered sessions.	9 months ago
George Joseph	6bc055416b	ARI: REST over Websocket ... This commit adds the ability to make ARI REST requests over the same websocket used to receive events. For full details on how to use the new capability, visit... https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/ Changes: * Added utilities to http.c: * ast_get_http_method_from_string(). * ast_http_parse_post_form(). * Added utilities to json.c: * ast_json_nvp_array_to_ast_variables(). * ast_variables_to_json_nvp_array(). * Added definitions for new events to carry REST responses. * Created res/ari/ari_websocket_requests.c to house the new request handlers. * Moved non-event specific code out of res/ari/resource_events.c into res/ari/ari_websockets.c * Refactored res/res_ari.c to move non-http code out of ast_ari_callback() (which is http specific) and into ast_ari_invoke() so it can be shared between both the http and websocket transports. UpgradeNote: This commit adds the ability to make ARI REST requests over the same websocket used to receive events. See https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/	9 months ago
Florent CHAUVEAU	ea657ec7c7	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,*,#...).	10 months ago
Norm Harrison	e8209bf56b	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>	10 months ago
Mark Murawski	abc8c5c93a	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	10 months ago
Alexei Gradinari	03cf8c62ad	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	10 months ago
Sean Bright	bae7dafa12	res_config_curl.c: Remove unnecessary warnings. ... Resolves: #1164	10 months ago
Sean Bright	3a9d7438e0	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	10 months ago
Joshua Elson	c4123901e5	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	10 months ago
Sean Bright	f685df5d14	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.	10 months ago
Maximilian Fridrich	a47fe8f84f	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.	10 months ago
Sean Bright	4dc2efc8c3	res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer. ... Found while reviewing #1128	10 months ago
George Joseph	46c9f7db8e	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	11 months ago
George Joseph	8976421504	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	11 months ago
George Joseph	71551013c4	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	11 months ago
George Joseph	5267c17645	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	11 months ago
Holger Hans Peter Freyther	71eb8a262f	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.	11 months ago
George Joseph	f1df1cacf6	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.	12 months ago
George Joseph	95b4b55b31	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	12 months ago
Sean Bright	b4156fecf0	docs: Indent <since> tags. ... Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	12 months ago
George Joseph	5c3b35bc79	res_pjsip: Fix startup/reload memory leak in config_auth. ... An issue in config_auth.c:ast_sip_auth_digest_algorithms_vector_init() was causing double allocations for the two supported_algorithms vectors to the tune of 915 bytes. The leak only happens on startup and when a reload is done and doesn't get bigger with the number of auth objects defined. * Pre-initialized the two vectors in config_auth:auth_alloc(). * Removed the allocations in ast_sip_auth_digest_algorithms_vector_init(). * Added a note to the doc for ast_sip_auth_digest_algorithms_vector_init() noting that the vector passed in should be initialized and empty. * Simplified the create_artificial_auth() function in pjsip_distributor. * Set the vector initialization count to 0 in config_global:global_apply().	12 months ago
George Joseph	85a4ab8390	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml	12 months ago
George Joseph	a47b8e2d40	docs: Add version information to manager event instance XML elements ... * Do a git blame on the embedded XML managerEvent elements. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The events defined in res/res_pjsip/pjsip_manager.xml were done by hand because the XML was extracted from the C source file relatively recently. Two bugs were fixed along the way... * The get_documentation awk script was exiting after it processed the first DOCUMENTATION block it found in a file. We have at least 1 source file with multiple DOCUMENTATION blocks so only the first one in them was being processed. The awk script was changed to continue searching rather than exiting after the first block. * Fixing the awk script revealed an issue in logger.c where the third DOCUMENTATION block contained a XML fragment that consisted only of a managerEventInstance element that wasn't wrapped in a managerEvent element. Since logger_doc.xml already existed, the remaining fragments in logger.c were moved to it and properly organized.	12 months ago
Sean Bright	fa286641fb	res_prometheus.c: Set Content-Type header on /metrics response. ... This should resolve the Prometheus error: > Error scraping target: non-compliant scrape target sending blank Content-Type and no fallback_scrape_protocol specified for target. Resolves: #1075	12 months ago
George Joseph	a22dc33057	docs: Add version information to configObject and configOption XML elements ... Most of the configObjects and configOptions that are implemented with ACO or Sorcery now have `<since>/<version>` elements added. There are probably some that the script I used didn't catch. The version tags were determined by the following... * Do a git blame on the API call that created the object or option. * From the commit hash, grab the summary line. * Do a `git log --grep <summary>` to find the cherry-pick commits in all branches that match. * Do a `git patch-id` to ensure the commits are all related and didn't get a false match on the summary. * Do a `git tag --contains <commit>` to find the tags that contain each commit. * Weed out all tags not <major>.<minor>.0. * Sort and discard any <major>.0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the API was last touched. configObjects and configOptions elements implemented with the base ast_config APIs were just not possible to find due to the non-deterministic way they are accessed. Also note that if the API call was on modified after it was added, the version will be the one it was last modified in. Final note: The configObject and configOption elements were introduced in 12.0.0 so options created before then may not have any XML documentation.	12 months ago
George Joseph	317b830c1e	res_pjsip_authenticator_digest: Fix issue with missing auth and DONT_OPTIMIZE ... The return code fom digest_check_auth wasn't explicitly being initialized. The return code also wasn't explicitly set to CHALLENGE when challenges were sent. When optimization was turned off (DONT_OPTIMIZE), the compiler was setting it to "0"(CHALLENGE) which worked fine. However, with optimization turned on, it was setting it to "1" (SUCCESS) so if there was no incoming Authorization header, the function was returning SUCCESS to the distributor allowing the request to incorrectly succeed. The return code is now initialized correctly and is now explicitly set to CHALLENGE when we send challenges.	12 months ago
George Joseph	3b53152624	docs: Various XML fixes ... * channels/pjsip/dialplan_functions_doc.xml: Added xmlns:xi to docs element. * main/bucket.c: Removed XML completely since the "bucket" and "file" objects are internal only with no config file. * main/named_acl.c: Fixed the configFile element name. It was "named_acl.conf" and should have been "acl.conf" * res/res_geolocation/geoloc_doc.xml: Added xmlns:xi to docs element. * res/res_http_media_cache.c: Fixed the configFile element name. It was "http_media_cache.conf" and should have been "res_http_media_cache.conf".	1 year ago
George Joseph	3e28ddce78	docs: Enable since/version handling for XML, CLI and ARI documentation ... * Added the "since" element to the XML configObject and configOption elements in appdocsxml.dtd. * Added the "Since" section to the following CLI output: ``` config show help <module> <object> config show help <module> <object> <option> core show application <app> core show function <func> manager show command <command> manager show event <event> agi show commands topic <topic> ``` * Refactored the commands above to output their sections in the same order: Synopsis, Since, Description, Syntax, Arguments, SeeAlso * Refactored the commands above so they all use the same pattern for writing the output to the CLI. * Fixed several memory leaks caused by failure to free temporary output buffers. * Added a "since" array to the mustache template for the top-level resources (Channel, Endpoint, etc.) and to the paths/methods underneath them. These will be added to the generated markdown if present. Example: ``` "resourcePath": "/api-docs/channels.{format}", "requiresModules": [ "res_stasis_answer", "res_stasis_playback", "res_stasis_recording", "res_stasis_snoop" ], "since": [ "18.0.0", "21.0.0" ], "apis": [ { "path": "/channels", "description": "Active channels", "operations": [ { "httpMethod": "GET", "since": [ "18.6.0", "21.8.0" ], "summary": "List all active channels in Asterisk.", "nickname": "list", "responseClass": "List[Channel]" }, ``` NOTE: No versioning information is actually added in this commit. Those will be added separately and instructions for adding and maintaining them will be published on the documentation site at a later date.	1 year ago
Artem Umerov	ee54dc56ee	logger.h: Fix build when AST_DEVMODE is not defined. ... Resolves: #1058	1 year ago
Sean Bright	7f13966202	manager: Add `<since>` tags for all AMI actions.	1 year ago
Abdelkader Boudih	1ea7d5cae6	res_config_pgsql: normalize database connection option with cel and cdr by supporting new options name	1 year ago
Stanislav Abramenkov	385e2d5042	res_pjproject: Fix typo (OpenmSSL->OpenSSL) ... Fix typo (OpenmSSL->OpenSSL) mentioned by bkford in #972	1 year ago
George Joseph	1933548d41	Add SHA-256 and SHA-512-256 as authentication digest algorithms ... * Refactored pjproject code to support the new algorithms and added a patch file to third-party/pjproject/patches * Added new parameters to the pjsip auth object: * password_digest = <algorithm>:<digest> * supported_algorithms_uac = List of algorithms to support when acting as a UAC. * supported_algorithms_uas = List of algorithms to support when acting as a UAS. See the auth object in pjsip.conf.sample for detailed info. * Updated both res_pjsip_authenticator_digest.c (for UAS) and res_pjsip_outbound_authentocator_digest.c (UAC) to suport the new algorithms. The new algorithms are only available with the bundled version of pjproject, or an external version > 2.14.1. OpenSSL version 1.1.1 or greater is required to support SHA-512-256. Resolves: #948 UserNote: The SHA-256 and SHA-512-256 algorithms are now available for authentication as both a UAS and a UAC.	1 year ago
Kent	85f5a047c1	res_pjsip: Add new AOR option "qualify_2xx_only" ... Added a new option "qualify_2xx_only" to the res_pjsip AOR qualify feature to mark a contact as available only if an OPTIONS request returns a 2XX response. If the option is not specified or is false, any response to the OPTIONS request marks the contact as available. UserNote: The pjsip.conf AOR section now has a "qualify_2xx_only" option that can be set so that only 2XX responses to OPTIONS requests used to qualify a contact will mark the contact as available.	1 year ago
Jaco Kroon	89ffbb5de7	res_odbc: release threads from potential starvation. ... Whenever a slot is freed up due to a failed connection, wake up a waiter before failing. In the case of a dead connection there could be waiters, for example, let's say two threads tries to acquire objects at the same time, with one in the cached connections, one will acquire the dead connection, and the other will enter into the wait state. The thread with the dead connection will clear up the dead connection, and then attempt a re-acquire (at this point there cannot be cached connections else the other thread would have received that and tried to clean up), as such, at this point we're guaranteed that either there are no waiting threads, or that the maxconnections - connection_cnt threads will attempt to re-acquire connections, and then either succeed, using those connections, or failing, and then signalling to release more waiters. Also fix the pointer log for ODBC handle %p dead which would always reflect NULL. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	1 year ago
Viktor Litvinov	607de36230	res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big ... Set Mark bit in rtp stream when timestamp skew is bigger than MAX_TIMESTAMP_SKEW. Fixes: #927	1 year ago
Alexey Vasilyev	9060a267e0	res_rtp_asterisk.c: Fix bridged_payload matching with sample rate for DTMF ... Fixes #1004	1 year ago
George Joseph	90cf13acd8	res_stir_shaken: Allow sending Identity headers for unknown TNs ... Added a new option "unknown_tn_attest_level" to allow Identity headers to be sent when a callerid TN isn't explicitly configured in stir_shaken.conf. Since there's no TN object, a private_key_file and public_cert_url must be configured in the attestation or profile objects. Since "unknown_tn_attest_level" uses the same enum as attest_level, some of the sorcery macros had to be refactored to allow sharing the enum and to/from string conversion functions. Also fixed a memory leak in crypto_utils:pem_file_cb(). Resolves: #921 UserNote: You can now set the "unknown_tn_attest_level" option in the attestation and/or profile objects in stir_shaken.conf to enable sending Identity headers for callerid TNs not explicitly configured.	1 year ago
George Joseph	ff7765a666	res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T ... The suppress_moh_on_sendonly endpoint option should have been defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES in the alembic script instead of OPT_YESNO_T and YESNO_VALUES. Also updated contrib/ast-db-manage/README.md to indicate that AST_BOOL_VALUES should always be used and provided an example. Resolves: #995	1 year ago
George Joseph	badf203203	res_pjsip: Add new endpoint option "suppress_moh_on_sendonly" ... Normally, when one party in a call sends Asterisk an SDP with a "sendonly" or "inactive" attribute it means "hold" and causes Asterisk to start playing MOH back to the other party. This can be problematic if it happens at certain times, such as in a 183 Progress message, because the MOH will replace any early media you may be playing to the calling party. If you set this option to "yes" on an endpoint and the endpoint receives an SDP with "sendonly" or "inactive", Asterisk will NOT play MOH back to the other party. Resolves: #979 UserNote: The new "suppress_moh_on_sendonly" endpoint option can be used to prevent playing MOH back to a caller if the remote end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.	1 year ago
Sean Bright	5c76486203	res_pjsip.c: Fix Contact header rendering for IPv6 addresses. ... Fix suggested by @nvsystems. Fixes #985	1 year ago
George Joseph	be8f3a3fa4	res_pjsip: Move tenantid to end of ast_sip_endpoint ... The tenantid field was originally added to the ast_sip_endpoint structure at the end of the AST_DECLARE_STRING_FIELDS block. This caused everything after it in the structure to move down in memory and break ABI compatibility. It's now at the end of the structure as an AST_STRING_FIELD_EXTENDED. Given the number of string fields in the structure now, the initial string field allocation was also increased from 64 to 128 bytes. Resolves: #982	1 year ago
Thomas Guebels	2ee258b0fc	pjsip_transport_events: handle multiple addresses for a domain ... The key used for transport monitors was the remote host name for the transport and not the remote address resolved for this domain. This was problematic for domains returning multiple addresses as several transport monitors were created with the same key. Whenever a subsystem wanted to register a callback it would always end up attached to the first transport monitor with a matching key. The key used for transport monitors is now the remote address and port the transport actually connected to. Fixes: #932	1 year ago
George Joseph	50bd50d798	res_srtp: Change Unsupported crypto suite msg from verbose to debug ... There's really no point in spamming logs with a verbose message for every unsupported crypto suite an older client may send in an SDP. If none are supported, there will be an error or warning.	1 year ago
Ben Ford	b8d818bd3c	Add res_pjsip_config_sangoma external module. ... Adds res_pjsip_config_sangoma as an external module that can be downloaded via menuselect. It lives under the Resource Modules section.	1 year ago
Thomas Guebels	6763dda90f	pjsip_transport_events: Avoid monitor destruction ... When a transport is disconnected, several events can arrive following each other. The first event will be PJSIP_TP_STATE_DISCONNECT and it will trigger the destruction of the transport monitor object. The lookup for the transport monitor to destroy is done using the transport key, that contains the transport destination host:port. A reconnect attempt by pjsip will be triggered as soon something needs to send a packet using that transport. This can happen directly after a disconnect since ca Subsequent events can arrive later like PJSIP_TP_STATE_DESTROY and will also try to trigger the destruction of the transport monitor if not already done. Since the lookup for the transport monitor to destroy is done using the transport key, it can match newly created transports towards the same destination and destroy their monitor object. Because of this, it was sometimes not possible to monitor a transport after one or more disconnections. This fix adds an additional check on the transport pointer to ensure only a monitor for that specific transport is removed. Fixes: #923	1 year ago