33744 Commits (6ebd820e269ec139e4a3ad8b5670df04a22c72df)

 

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Author	SHA1	Message	Date
George Joseph	cd2865175c	.github: Fix quoting in PROpenedOrUpdated	2 years ago
George Joseph	7c917618f4	.github: Add cherry-pick reminder to new PRs	2 years ago
Jaco Kroon	0e6295128c	configure: fix test code to match gethostbyname_r prototype. (#75) ... This enables the test to work with CC=clang. Without this the test for 6 args would fail with: utils.c:99:12: error: static declaration of 'gethostbyname_r' follows non-static declaration static int gethostbyname_r (const char *name, struct hostent *ret, char *buf, ^ /usr/include/netdb.h:177:12: note: previous declaration is here extern int gethostbyname_r (const char *__restrict __name, ^ Fixing the expected return type to int sorts this out. Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
Sean Bright	ae6b56e357	res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#47) ... The functionality we are interested in is present only in pjsip 2.13 and newer. Resolves: #45	2 years ago
zhengsh	c8ce2c705d	res_sorcery_memory_cache.c: Fix memory leak (#56) ... Replace the original call to ast_strdup with a call to ast_strdupa to fix the leak issue. Resolves: #55 ASTERISK-30429	2 years ago
Sean Bright	f414815159	utils.h: Deprecate `ast_gethostbyname()`. (#79) ... Deprecate `ast_gethostbyname()` in favor of `ast_sockaddr_resolve()` and `ast_sockaddr_resolve_first_af()`. `ast_gethostbyname()` has not been used by any in-tree code since 2021. This function will be removed entirely in Asterisk 23. Resolves: #78 UpgradeNote: ast_gethostbyname() has been deprecated and will be removed in Asterisk 23. New code should use `ast_sockaddr_resolve()` and `ast_sockaddr_resolve_first_af()`.	2 years ago
Sean Bright	d59a8ef59e	xml.c: Process XML Inclusions recursively. (#69) ... If processing an XInclude results in new <xi:include> elements, we need to run XInclude processing again. This continues until no replacement occurs or an error is encountered. There is a separate issue with dynamic strings (ast_str) that will be addressed separately. Resolves: #65	2 years ago
Joshua C. Colp	172a1a9d0c	.github: Tweak improvement issue type language.	2 years ago
Gitea	60ca49d288	.github: Tweak new feature language, and move feature requests elsewhere.	2 years ago
Joshua C. Colp	eec85c672c	.github: Fix staleness check to only run on certain labels.	2 years ago
George Joseph	857bc88d14	.github: Add AsteriskReleaser	2 years ago
Henning Westerholt	1a7866b172	chan_pjsip: also return all codecs on empty re-INVITE for late offers (#59) ... We should also return all codecs on an re-INVITE without SDP for a call that used late offer (e.g. no SDP in the initial INVITE, SDP in the ACK). Bugfix for feature introduced in ASTERISK-30193 (https://issues.asterisk.org/jira/browse/ASTERISK-30193) Migration from previous gerrit change that was not merged.	2 years ago
Mike Bradeen	cd48733353	cel: add local optimization begin event (#54) ... The current AST_CEL_LOCAL_OPTIMIZE event is and has been triggered on a local optimization end to serve as a flag indicating the event occurred. This change adds a second AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail. Resolves: #52 UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired. UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.	2 years ago
Sean Bright	0d6b271831	core: Cleanup gerrit and JIRA references. (#58) ... * Remove .gitreview and switch to pulling the main asterisk branch version from configure.ac instead. * Replace references to JIRA with GitHub. * Other minor cleanup found along the way. Resolves: #39	2 years ago
George Joseph	f5168354cb	.github: Fix CherryPickTest to only run when it should ... Fixed CherryPickTest so it triggers only on the "cherry-pick-test" label instead of all labels.	2 years ago
George Joseph	676cfc3d5a	.github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS	2 years ago
George Joseph	79a383ebc6	.github: Remove separate set labels step from new PR	2 years ago
George Joseph	06c6a8b064	.github: Refactor CP progress and add new PR test progress	2 years ago
Maximilian Fridrich	cacd98bb29	res_pjsip: mediasec: Add Security-Client headers after 401 (#49) ... When using mediasec, requests sent after a 401 must still contain the Security-Client header according to draft-dawes-sipcore-mediasec-parameter. Resolves: #48	2 years ago
George Joseph	65fa8d6009	.github: Add cherry-pick test progress labels	2 years ago
Joshua C. Colp	7526d1d6c1	LICENSE: Update link to trademark policy. (#44) ... Resolves: #43	2 years ago
InterLinked1	ffb90c4549	say.c: Fix French time playback. (#42) ... ast_waitstream was not called after ast_streamfile, resulting in "o'clock" being skipped in French. Additionally, the minute announcements should be feminine. Reported-by: Danny Lloyd Resolves: #41 ASTERISK-30488	2 years ago
Naveen Albert	9a999242b2	chan_dahdi: Add dialmode option for FXS lines. ... Currently, both pulse and tone dialing are always enabled on all FXS lines, with no way of disabling one or the other. In some circumstances, it is desirable or necessary to disable one of these, and this behavior can be problematic. A new "dialmode" option is added which allows setting the methods to support on a per channel basis for FXS (FXO signalled lines). The four options are "both", "pulse", "dtmf"/"tone", and "none". Additionally, integration with the CHANNEL function is added so that this setting can be updated for a channel during a call. Resolves: #35 ASTERISK-29992 UserNote: A "dialmode" option has been added which allows specifying, on a per-channel basis, what methods of subscriber dialing (pulse and/or tone) are permitted. Additionally, this can be changed on a channel at any point during a call using the CHANNEL function.	2 years ago
George Joseph	1b95f6ee3f	.github: Update issue templates	2 years ago
George Joseph	11ce72fc8a	.github: Remove unnecessary parameter in CherryPickTest	2 years ago
George Joseph	f50cc8852d	Initial GitHub PRs	2 years ago
George Joseph	eaec5a35dc	Initial GitHub Issue Templates	2 years ago
Joshua C. Colp	21c07cf6e1	pbx_dundi: Fix PJSIP endpoint configuration check. ... ASTERISK-28233 Change-Id: I0f11c096b307a6178e22ca49d9c756343f0e1fdc	2 years ago
Joshua Colp	4ec4543332	Revert "app_queue: periodic announcement configurable start time." ... This reverts commit 71e317f68f. Reason for revert: Causes segmentation fault. Change-Id: I3beeda83249bffec2a8f246aa50a6b2f1b59ef59	2 years ago
Naveen Albert	0119f3ad48	res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters. ... The current STIR/SHAKEN signing process is inconsistent with the RFCs in a couple ways that can cause interoperability issues. RFC8225 specifies that the keys must be ordered lexicographically, but currently the fields are simply ordered according to the order in which they were added to the JSON object, which is not compliant with the RFC and can cause issues with some carriers. To fix this, we now leverage libjansson's ability to dump a JSON object sorted by key value, yielding the correct field ordering. Additionally, telephone numbers must have any leading + prefix removed and must not contain characters outside of 0-9, *, and # in order to comply with the RFCs. Numbers are now properly formatted as such. ASTERISK-30407 #close Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2	2 years ago
Naveen Albert	ecf49ff746	pbx_dundi: Add PJSIP support. ... Adds PJSIP as a supported technology to DUNDi. To facilitate this, we now allow an endpoint to be specified for outgoing PJSIP calls. We also allow users to force a specific channel technology for outgoing SIP-protocol calls. ASTERISK-28109 #close ASTERISK-28233 #close Change-Id: I2e28e5a5d007bd49e3df113ad567b011103899bf	2 years ago
Henning Westerholt	58e88dec90	chan_pjsip: fix music on hold continues after INVITE with replaces ... In a three party scenario with INVITE with replaces, we need to unhold the call, otherwise one party continues to get music on hold, and the call is not properly bridged between them. ASTERISK-30428 Change-Id: I5675df11e739be5226b328f8828d4b8d81fbefb4	2 years ago
The_Blode	0c50ab0d4f	install_prereq: Add Linux Mint support. ... ASTERISK-30359 #close Change-Id: I9c140c7f12ca7dafe65c317f2a26122cf2c72556	2 years ago
Naveen Albert	eadf28a476	voicemail.conf: Fix incorrect comment about #include. ... A comment at the top of voicemail.conf says that #include cannot be used in voicemail.conf because this breaks the ability for app_voicemail to auto-update passwords. This is factually incorrect, since Asterisk has no problem updating files that are #include'd in the main configuration file, and this does work in voicemail.conf as well. ASTERISK-30479 #close Change-Id: I3bf7d275849ab83f55f7fb6702a75a3077ee1df3	2 years ago
Naveen Albert	fa41458298	app_queue: Fix minor xmldoc duplication and vagueness. ... The F option in the xmldocs for the Queue application was erroneously duplicated, causing it to display twice on the wiki. The two sections are now merged into one. Additionally, the description for the d option was quite vague. Some more details are added to provide context as to what this actually does. ASTERISK-30486 #close Change-Id: I6706cea708b5cc781f59f8652c2cb377e55aed7e	2 years ago
George Joseph	1245e3330b	test.c: Fix counting of tests and add 2 new tests ... The unit test XML output was counting all registered tests as "run" even when only a subset were actually requested to be run and the "failures" attribute was missing. * The "tests" attribute of the "testsuite" element in the output XML now reflects only the tests actually requested to be executed instead of all the tests registered. * The "failures" attribute was added to the "testsuite" element. Also added 2 new unit tests that just pass and fail to be used for CI testing. Change-Id: Ia137814b5aeb0e1a44c75034bd3615c26021da69	2 years ago
Mike Bradeen	e494a55467	res_pjsip_pubsub: subscription cleanup changes ... There are two main parts of the change associated with this commit. These are driven by the change in call order of pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject when an in-dialog SUBSCRIBE is received. First, the previous behavior was for pjproject to call pubsub_on_rx_refresh before calling pubsub_on_evsub_state when an in-dialog SUBSCRIBE was received that changes the subscription state. If that change was a termination due to a re-SUBSCRIBE with an expires of 0, we used to use the call to pubsub_on_rx_refresh to set the substate of the evsub to TERMINATE_PENDING before pjproject could call pubsub_on_evsub_state. This substate let pubsub_on_evsub_state know that the subscription TERMINATED event could be ignored as there was still a subsequent NOTIFY that needed to be generated and another call to pubsub_on_evsub_state to come with it. That NOTIFY was sent via serialized_pubsub_on_refresh_timeout which would see the TERMINATE_PENDING state and transition it to TERMINATE_IN_PROGRESS before triggering another call to pubsub_on_evsub_state (which now would clean up the evsub.) The new pjproject behavior is to call pubsub_on_evsub_state before pubsub_on_rx_refresh. This means we no longer can set the state to TERMINATE_PENDING to tell pubsub_on_evsub_state that it can ignore the first TERMINATED event. To handle this, we now look directly at the event type, method type and the expires value to determine whether we want to ignore the event or use it to trigger the evsub cleanup. Second, pjproject now expects the NOTIFY to actually be sent during pubsub_on_rx_refresh and avoids the protocol violation inherent in sending a NOTIFY before the SUBSCRIBE is acknowledged by caching the sent NOTIFY then sending it after responding to the SUBSCRIBE. This requires we send the NOTIFY using the non-serialized pubsub_on_refresh_timeout directly and let pjproject handle the protocol violation. ASTERISK-30469 Change-Id: I05c1d91a44fe28244ae93faa4a2268a3332b5fd7	2 years ago
Jaco Kroon	3d86701a86	res_calendar: output busy state as part of show calendar. ... Change-Id: I894e4ecc3e93db4ff7783d46266ba3c5e6ccda10 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
Sean Bright	acd1513111	ael: Regenerate lexers and parsers. ... Various changes to ensure that the lexers and parsers can be correctly generated when REBUILD_PARSERS is enabled. Some notes: * Because of the version of flex we are using to generate the lexers (2.5.35) some post-processing in the Makefile is still required. * The generated lexers do not contain the problematic C99 check that was being replaced by the call to sed in the respective Makefiles so it was removed. * Since these files are generated, they will include trailing whitespace in some places. This does not need to be corrected. Change-Id: Ibbd343606fcf5c0d285b1599e6e8e59f514f2e4e	2 years ago
Sean Bright	7dc73c8151	loader.c: Minor module key check simplification. ... Change-Id: I65aefd4434a783096165c179b5f94f2e4810dffe	2 years ago
Sean Bright	03e7bbbce9	Revert "pbx_ael: Global variables are not expanded." ... This reverts commit c448dcd2f0. Reason for revert: Behavior change that breaks existing dialplan. ASTERISK-30472 #close Change-Id: I20e44b4081d6ee0fe54cde44ac71dcf2d146f909	2 years ago
Mike Bradeen	edd7f1b060	bridge_builtin_features: add beep via touch variable ... Add periodic beep option to one-touch recording by setting the touch variable TOUCH_MONITOR_BEEP or TOUCH_MIXMONITOR_BEEP to the desired interval in seconds. If the interval is less than 5 seconds, a minimum of 5 seconds will be imposed. If the interval is set to an invalid value, it will default to 15 seconds. A new test event PERIODIC_HOOK_ENABLED was added to the func_periodic_hook hook_on function to indicate when a hook is started. This is so we can test that the touch variable starts the hook as expected. ASTERISK-30446 Change-Id: I800e494a789ba7a930bbdcd717e89d86040d6661	2 years ago
Mike Bradeen	c4a55322bc	res_mixmonitor: MixMonitorMute by MixMonitor ID ... While it is possible to create multiple mixmonitor instances on a channel, it was not previously possible to mute individual instances. This change includes the ability to specify the MixMonitorID when calling the manager action: MixMonitorMute. This will allow an individual MixMonitor instance to be muted via id. This id can be stored as a channel variable using the 'i' MixMonitor option. As part of this change, if no MixMonitorID is specified in the manager action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor spy-type audiohooks on the channel. This is done via the new audiohook function: ast_audiohook_set_mute_all. ASTERISK-30464 Change-Id: Ibba8c7e750577aa1595a24b23316ef445245be98	2 years ago
Mike Bradeen	9504d17ad2	format_sln: add .slin as supported file extension ... Adds '.slin' to existing supported file extensions: .sln and .raw ASTERISK-30465 Change-Id: Ice848addc03a64c8404b87cb5d3b13399c57e496	2 years ago
Mike Bradeen	76da3c71a7	cli: increase channel column width ... For 'core show channels', the Channel name field is increased to 64 characters and the Location name field is increased to 32 characters. For 'core show channels verbose', the Channel name field is increased to 80 characters, the Context is increased to 24 characters and the Extension is increased to 24 characters. ASTERISK-30455 Change-Id: Ibec3742ce360ffc93bc56e9984c2a21dabc4d5e1	2 years ago
Jaco Kroon	71e317f68f	app_queue: periodic announcement configurable start time. ... This newly introduced periodic-announce-startdelay makes it possible to configure the initial start delay of the first periodic announcement after which periodic-announce-frequency takes over. ASTERISK-30437 #close Change-Id: Ia79984b6377ef78f167ad9ea2ac084bec29955d0 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	2 years ago
Naveen Albert	907692abb3	app_osplookup: Remove obsolete sample config. ... ASTERISK_30302 previously removed app_osplookup, but its sample config was not removed. This removes it since nothing else uses it. ASTERISK-30438 #close Change-Id: Ife234208f5f197644475db4ab1af95a8551642fd	2 years ago
Naveen Albert	ff6c293ec0	func_json: Fix JSON parsing issues. ... Fix issue with returning empty instead of dumping the JSON string when recursing. Also adds a unit test to capture this fix. ASTERISK-30441 #close Change-Id: If0bde9f3fe84f7af485e0838205cc21e0f752a85	2 years ago
Naveen Albert	95a41d231f	app_dial: Fix DTMF not relayed to caller on unanswered calls. ... DTMF frames are not handled in app_dial when sent towards the caller. This means that if DTMF is sent to the calling party and the call has not yet been answered, the DTMF is not audible. This is now fixed by relaying DTMF frames if only a single destination is being dialed. ASTERISK-29516 #close Change-Id: Iafd7430ac2915126d42dc48f0b73b262452ee027	2 years ago
Fabrice Fontaine	faf58ccc70	configure: fix detection of re-entrant resolver functions ... uClibc does not provide res_nsearch: asterisk-16.0.0/main/dns.c:506: undefined reference to `res_nsearch' Patch coded by Yann E. MORIN: http://lists.busybox.net/pipermail/buildroot/2018-October/232630.html ASTERISK-21795 #close Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de> [Retrieved from: https: //git.buildroot.net/buildroot/tree/package/asterisk/0005-configure-fix-detection-of-re-entrant-resolver-funct.patch] Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com> Change-Id: I79296f19e28ec764bbd1e991bf11c416d0b10563	2 years ago