asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	e0edea0383	app_voicemail: Fix ill-formatted pager emails with custom subject. Add missing end-of-headers newline to pager emails with custom subjects, since this was missing from this code path. Resolves: #902 (cherry picked from commit `e63da10a17`)	11 months ago
Sean Bright	4ade5a21b7	res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery Fixes #895 (cherry picked from commit `dad84444fc`)	11 months ago
George Joseph	6dc00f642a	Fix application references to Background The app is actually named "BackGround" but several references in XML documentation were spelled "Background" with the lower case "g". This was causing documentation links to return "not found" messages. (cherry picked from commit `f71668f0b9`)	11 months ago
George Joseph	82a229166a	manager.conf.sample: Fix mathcing typo (cherry picked from commit `adaacc281d`)	11 months ago
George Joseph	56d3070939	manager: Enhance event filtering for performance UserNote: You can now perform more granular filtering on events in manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/` This is much more efficient than `eventfilter = Event: Newchannel.*Channel: PJSIP/` Full syntax guide is in configs/samples/manager.conf.sample. (cherry picked from commit `b8d16fd800`)	11 months ago
George Joseph	2ca93f0cdc	manager.c: Split XML documentation to manager_doc.xml (cherry picked from commit `dd643bf827`)	11 months ago
George Joseph	03883ccb8e	.github: Fix realtime param on Weekly and Nightly tests and... Rename the "Cleanup" job in the cherry-pick and recheck jobs to "Summary". (cherry picked from commit `0c5a114b9f`)	11 months ago
George Joseph	5e9ea95517	.github: Add WeeklyTests and make Nightlies Monday-Saturday ...and add "realtime" option. (cherry picked from commit `d17bf1af1a`)	11 months ago
George Joseph	6297462fd2	db.c: Remove limit on family/key length Consumers like media_cache have been running into issues with the previous astdb "/family/key" limit of 253 bytes when needing to store things like long URIs. An Amazon S3 URI is a good example of this. Now, instead of using a static 256 byte buffer for "/family/key", we use ast_asprintf() to dynamically create it. Both test_db.c and test_media_cache.c were also updated to use keys/URIs over the old 253 character limit. Resolves: #881 UserNote: The `ast_db_()` APIs have had the 253 byte limit on "/family/key" removed and will now accept families and keys with a total length of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB` dialplan applications, dialplan functions, manager actions and `databse` CLI commands. Since the media_cache also uses the `ast_db_*()` APIs, you can now store resources with URIs longer than 253 bytes. (cherry picked from commit `ff7d094803`)	11 months ago
George Joseph	7788fac511	.github: Changes required to use cached builds and shorten names (cherry picked from commit `6c0fa2274b`)	11 months ago
Asterisk Development Team	8e4a09f711	Update for 22.0.0	12 months ago
Asterisk Development Team	4473ed5256	Update for 22.0.0-rc2	1 year ago
George Joseph	fd8d28cb60	stir_shaken: Fix propagation of attest_level and a few other values attest_level, send_mky and check_tn_cert_public_url weren't propagating correctly from the attestation object to the profile and tn. * In the case of attest_level, the enum needed to be changed so the "0" value (the default) was "NOT_SET" instead of "A". This now allows the merging of the attestation object, profile and tn to detect when a value isn't set and use the higher level value. * For send_mky and check_tn_cert_public_url, the tn default was forced to "NO" which always overrode the profile and attestation objects. Their defaults are now "NOT_SET" so the propagation happens correctly. * Just to remove some redundant code in tn_config.c, a bunch of calls to generate_sorcery_enum_from_str() and generate_sorcery_enum_to_str() were replaced with a single call to generate_acfg_common_sorcery_handlers(). Resolves: #904	1 year ago
George Joseph	dc8f83662b	res_stir_shaken: Remove stale include for jansson.h in verification.c verification.c had an include for jansson.h left over from previous versions of the module. Since res_stir_shaken no longer has a dependency on jansson, the bundled version wasn't added to GCC's include path so if you didn't also have a jansson development package installed, the compile would fail. Removing the stale include was the only thing needed. Resolves: #889	1 year ago
George Joseph	4411f22d67	res_stir_shaken.c: Fix crash when stir_shaken.conf is invalid * If the call to ast_config_load() returns CONFIG_STATUS_FILEINVALID, check_for_old_config() now returns LOAD_DECLINE instead of continuing on with a bad pointer. * If CONFIG_STATUS_FILEMISSING is returned, check_for_old_config() assumes the config is being loaded from realtime and now returns LOAD_SUCCESS. If it's actually not being loaded from realtime, sorcery will catch that later on. * Also refactored the error handling in load_module() a bit. Resolves: #884	1 year ago
Asterisk Development Team	7c1687ba12	Update for 22.0.0-rc1	1 year ago
George Joseph	3fb205f3af	res_stir_shaken: Check for disabled before param validation For both attestation and verification, we now check whether they've been disabled either globally or by the profile before validating things like callerid, orig_tn, dest_tn, etc. This prevents useless error messages. Resolves: #879 (cherry picked from commit `ef7d177ee3`)	1 year ago
Tinet-mucw	ef2bae8464	app_chanspy.c: resolving the issue writing frame to whisper audiohook. ChanSpy(${channel}, qEoSw): because flags set o, ast_audiohook_set_frame_feed_direction(audiohook, AST_AUDIOHOOK_DIRECTION_READ); this will effect whisper audiohook and spy audiohook, this makes writing frame to whisper audiohook impossible. So add function start_whispering to starting whisper audiohook. Resolves: #876 (cherry picked from commit `ffb1dca485`)	1 year ago
Alexei Gradinari	a06c18e8bd	autoservice: Do not sleep if autoservice_stop is called within autoservice thread It's possible that ast_autoservice_stop is called within the autoservice thread. In this case the autoservice thread is stuck in an endless sleep. To avoid endless sleep ast_autoservice_stop must check that it's not called within the autoservice thread. Fixes: #763 (cherry picked from commit `6c5c1e5426`)	1 year ago
George Joseph	b506faaa2d	res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback The ub_result pointer passed to unbound_resolver_callback by libunbound can be NULL if the query was for something malformed like `.1` or `[.1]`. If it is, we now set a 'ns_r_formerr' result and return instead of crashing with a SEGV. This causes pjproject to simply cancel the transaction with a "No answer record in the DNS response" error. The existing "off nominal" unit test was also updated to check this condition. Although not necessary for this fix, we also made ast_dns_resolver_completed() tolerant of a NULL result. Resolves: GHSA-v428-g3cw-7hv9 (cherry picked from commit `d2d0c507ff`)	1 year ago
George Joseph	78f8b00267	app_voicemail: Use ast_asprintf to create mailbox SQL query ...instead of trying to calculate the length of the buffer needed manually. (cherry picked from commit `dc97763979`)	1 year ago
Mike Bradeen	add94582ba	res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch When Asterisk sends an offer to Bob that includes 48K and 8K codecs with matching 4733 offers, Bob may want to use the 48K audio codec but can not accept 48K digits and so negotiates for a mixed set. Asterisk will now check Bob's offer to make sure Bob has indicated this is acceptible and if not, will use Bob's preference. Fixes: #847 (cherry picked from commit `4972cc8668`)	1 year ago
Tinet-mucw	b42d4f8a3c	app_chanspy.c: resolving the issue with audiohook direction read ChanSpy(${channel}, qEoS): When chanspy spy the direction read, reading frame is often failed when reading direction read audiohook. because chanspy only read audiohook direction read; write_factory_ms will greater than 100ms soon, then ast_slinfactory_flush will being called, then direction read will fail. Resolves: #861 (cherry picked from commit `f646727423`)	1 year ago
George Joseph	cd8ffa1225	security_agreements.c: Refactor the to_str functions and fix a few other bugs * A static array of security mechanism type names was created. * ast_sip_str_to_security_mechanism_type() was refactored to do a lookup in the new array instead of using fixed "if/else if" statments. * security_mechanism_to_str() and ast_sip_security_mechanisms_to_str() were refactored to use ast_str instead of a fixed length buffer to store the result. * ast_sip_security_mechanism_type_to_str was removed in favor of just referencing the new type name array. Despite starting with "ast_sip_", it was a static function so removing it doesn't affect ABI. * Speaking of "ast_sip_", several other static functions that started with "ast_sip_" were renamed to avoid confusion about their public availability. * A few VECTOR free loops were replaced with AST_VECTOR_RESET(). * Fixed a meomry leak in pjsip_configuration.c endpoint_destructor caused by not calling ast_sip_security_mechanisms_vector_destroy(). * Fixed a memory leak in res_pjsip_outbound_registration.c add_security_headers() caused by not specifying OBJ_NODATA in an ao2_callback. * Fixed a few ao2_callback return code misuses. Resolves: #845 (cherry picked from commit `efe34dc6e5`)	1 year ago
Alexei Gradinari	045d6c5fc1	res_pjsip_sdp_rtp fix leaking astobj2 ast_format PR #700 added a preferred_format for the struct ast_rtp_codecs, but when set the preferred_format it leaks an astobj2 ast_format. In the next code ast_rtp_codecs_set_preferred_format(&codecs, ast_format_cap_get_format(joint, 0)); both functions ast_rtp_codecs_set_preferred_format and ast_format_cap_get_format increases the ao2 reference count. Fixes: #856 (cherry picked from commit `6ef4548223`)	1 year ago
George Joseph	89bdcf0112	stir_shaken.conf.sample: Fix bad references to private_key_path They should be private_key_file. Resolves: #854 (cherry picked from commit `ca15a2c034`)	1 year ago
Sean Bright	6b7b5c98df	res_pjsip_logger.c: Fix 'OPTIONS' tab completion. Fixes #843 (cherry picked from commit `b0e1f6a6ce`)	1 year ago
Sean Bright	5a1e6995d6	alembic: Make 'revises' header comment match reality. (cherry picked from commit `53f2c426cc`)	1 year ago
Mike Bradeen	31b8dc0923	Update version for Asterisk 22	1 year ago
Cade Parker	7f2ac6cb4f	chan_mobile: decrease CHANNEL_FRAME_SIZE to prevent delay On modern Bluetooth devices or lower-powered asterisk servers, decreasing the channel frame size significantly improves latency and delay on outbound calls with only a mild sacrifice to the quality of the call (the frame size before was massive overkill to begin with)	1 year ago
Mike Bradeen	b2455b2732	res_pjsip_notify: add dialplan application Add dialplan application PJSIPNOTIFY to send either pre-configured NOTIFY messages from pjsip_notify.conf or with headers defined in dialplan. Also adds the ability to send pre-configured NOTIFY commands to a channel via the CLI. Resolves: #799 UserNote: A new dialplan application PJSIPNotify is now available which can send SIP NOTIFY requests from the dialplan. The pjsip send notify CLI command has also been enhanced to allow sending NOTIFY messages to a specific channel. Syntax: pjsip send notify <option> channel <channel>	1 year ago
George Joseph	2328a0ab3a	manager.c: Fix FRACK when doing CoreShowChannelMap in DEVMODE If you run an AMI CoreShowChannelMap on a channel that isn't in a bridge and you're in DEVMODE, you can get a FRACK because the bridge id is empty. We now simply return an empty list for that request.	1 year ago
Ben Ford	fb0e46e464	channel: Add multi-tenant identifier. This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.	1 year ago
Jaco Kroon	3f1d1a1463	configure: Use . file rather than source file. source is a bash concept, so when /bin/sh points to another shell the existing construct won't work. Reference: https://bugs.gentoo.org/927055 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	1 year ago
gibbz00	99a5064a07	feat: ARI "ChannelToneDetected" event A stasis event is now produced when using the TONE_DETECT dialplan function. This event is published over ARI using the ChannelToneDetected event. This change does not make it available over AMI. Fixes: #811 UserNote: Setting the TONE_DETECT dialplan function on a channel in ARI will now cause a ChannelToneDetected ARI event to be raised when the specified tone is detected.	1 year ago
George Joseph	0f67c2696f	manager.c: Add entries to Originate blacklist Added Reload and DBdeltree to the list of dialplan application that can't be executed via the Originate manager action without also having write SYSTEM permissions. Added CURL, DB, FILE, ODBC and REALTIME to the list of dialplan functions that can't be executed via the Originate manager action without also having write SYSTEM permissions. If the Queue application is attempted to be run by the Originate manager action and an AGI parameter is specified in the app data, it'll be rejected unless the manager user has either the AGI or SYSTEM permissions. Resolves: #GHSA-c4cg-9275-6w44	1 year ago
Mike Bradeen	a824a0cdb4	res_stasis: fix intermittent delays on adding channel to bridge Previously, on command execution, the control thread was awoken by sending a SIGURG. It was found that this still resulted in some instances where the thread was not immediately awoken. This change instead sends a null frame to awaken the control thread, which awakens the thread more consistently. Resolves: #801	1 year ago
Tinet-mucw	291c1e4c1f	res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto When the endpoint dtmf_mode is set to auto, a SIP request is sent to the UAC, and the SIP SDP from the UAC does not include the telephone-event. Later, the UAC sends an INVITE, and the SIP SDP includes the telephone-event. In this case, DTMF should be sent by RFC2833 rather than using inband signaling. Resolves: asterisk#826	1 year ago
George Joseph	b7879224c1	.github: Allow testing an Asterisk PR against a testsuite PR	1 year ago
George Joseph	e4b0ad45e9	.github: Add params to Releaser for FPBX issue creation	1 year ago
George Joseph	cee50bab7a	rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset() There can be empty slots in payload_mapping_tx corresponding to dynamic payload types that haven't been seen before so we now check for NULL before attempting to use 'type' in the call to ast_format_cmp. Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset() Resolves: #822	1 year ago
George Joseph	a02fc685a8	stir_shaken: CRL fixes and a new CLI command * Fixed a bug in crypto_show_cli_store that was causing asterisk to crash if there were certificate revocation lists in the verification certificate store. We're also now prefixing certificates with "Cert:" and CRLs with "CRL:" to distinguish them in the list. * Added 'untrusted_cert_file' and 'untrusted_cert_path' options to both verification and profile objects. If you have CRLs that are signed by a different CA than the incoming X5U certificate (indirect CRL), you'll need to provide the certificate of the CRL signer here. Thse will show up as 'Untrusted" when showing the verification or profile objects. * Fixed loading of crl_path. The OpenSSL API we were using to load CRLs won't actually load them from a directory, only a file. We now scan the directory ourselves and load the files one-by-one. * Fixed the verification flags being set on the certificate store. - Removed the CRL_CHECK_ALL flag as this was causing all certificates to be checked for CRL extensions and failing to verify the cert if there was none. This basically caused all certs to fail when a CRL was provided via crl_file or crl_path. - Added the EXTENDED_CRL_SUPPORT flag as it is required to handle indirect CRLs. * Added a new CLI command... `stir_shaken verify certificate_file <certificate_file> [ <profile> ]` which will assist troubleshooting certificate problems by allowing the user to manually verify a certificate file against either the global verification certificate store or the store for a specific profile. * Updated the XML documentation and the sample config file. Resolves: #809	1 year ago
George Joseph	f45f8781d0	res_pjsip_config_wizard.c: Refactor load process The way we have been initializing the config wizard prevented it from registering its objects if res_pjsip happened to load before it. * We now use the object_type_registered sorcery observer to kick things off instead of the wizard_mapped observer. * The load_module function now checks if res_pjsip has been loaded already and if it was it fires the proper observers so the objects load correctly. Resolves: #816 UserNote: The res_pjsip_config_wizard.so module can now be reloaded.	1 year ago
George Joseph	79e0b50162	voicemail.conf.sample: Fix ':' comment typo ...and removed an errant trailing space. Resolves: #819	1 year ago
George Joseph	bc05416004	bridge_softmix: Fix queueing VIDUPDATE control frames softmix_bridge_write_control() now calls ast_bridge_queue_everyone_else() with the bridge_channel so the VIDUPDATE control frame isn't echoed back. softmix_bridge_write_control() was setting bridge_channel to NULL when calling ast_bridge_queue_everyone_else() for VIDUPDATE control frames. This was causing the frame to be echoed back to the channel it came from. In certain cases, like when two channels or bridges are being recorded, this can cause a ping-pong effect that floods the system with VIDUPDATE control frames. Resolves: #780	1 year ago
Igor Goncharovsky	55f4e6dd03	res_pjsip_path.c: Fix path when dialing using PJSIP_DIAL_CONTACTS() When using the PJSIP_DIAL_CONTACTS() function for use in the Dial() command, the contacts are returned in text form, so the input to the path_outgoing_request() function is a contact value of NULL. The issue was reported in ASTERISK-28211, but was not actually fixed in ASTERISK-30100. This fix brings back the code that was previously removed and adds code to search for a contact to extract the path value from it.	1 year ago
Mike Bradeen	52bdfe09ab	res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits After change made in `624f509` to add support for non 8K RFC 4733/2833 digits, Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of the negotiated codec(s). This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec. A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to be re-written to allow for these scenarios. Fixes: #776	1 year ago
George Joseph	c0d8831636	.github: Pass app_id and app_priv_key to AsteriskMergePR	1 year ago
George Joseph	6cce75409e	.github: Change OnPRMergeApproved to use default token	1 year ago
George Joseph	e6633e8fdd	ast-db-manage: Remove duplicate enum creation Remove duplicate creation of ast_bool_values from 2b7c507d7d12_add_queue_log_option_log_restricted_.py. This was causing alembic upgrades to fail since the enum was already created in fe6592859b85_fix_mwi_subscribe_replaces_.py back in 2018. Resolves: #797	1 year ago

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