asterisk

Commit Graph

Author	SHA1	Message	Date
Gitea	a6c742dbf6	.github: Tweak new feature language, and move feature requests elsewhere. (cherry picked from commit `e1b29b636c`)	2 years ago
Joshua C. Colp	5723db8a44	.github: Fix staleness check to only run on certain labels. (cherry picked from commit `d7b647e97f`)	2 years ago
Asterisk Development Team	c0d6dac258	Update for 18.18.1	2 years ago
Sean Bright	7b55312ca0	apply_patches: Use globbing instead of file/sort. This accomplishes the same thing as a `find ... \| sort` but with the added benefit of clarity and avoiding a call to a subshell. Additionally drop the -s option from call to patch as it is not POSIX.	2 years ago
George Joseph	f0a6e84f16	apply_patches: Sort patch list before applying The apply_patches script wasn't sorting the list of patches in the "patches" directory before applying them. This left the list in an indeterminate order. In most cases, the list is actually sorted but rarely, they can be out of order and cause dependent patches to fail to apply. We now sort the list but the "sort" program wasn't in the configure scripts so we needed to add that and regenerate the scripts as well. Resolves: #193	2 years ago
Stanislav Abramenkov	8c31fd2de3	pjsip: Upgrade bundled version to pjproject 2.13.1	2 years ago
Asterisk Development Team	e08f691e28	Update for 18.18.0	2 years ago
George Joseph	83418725e7	Set up new ChangeLogs directory	2 years ago
Asterisk Development Team	00d6a999d6	Update for 18.18.0-rc1	2 years ago
George Joseph	3976b1997c	.github: Add AsteriskReleaser (cherry picked from commit `2392a5bb74`)	2 years ago
Henning Westerholt	ebee6f38f0	chan_pjsip: also return all codecs on empty re-INVITE for late offers 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. (cherry picked from commit `8711c84e3c`)	2 years ago
Mike Bradeen	1cba481518	cel: add local optimization begin event 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. (cherry picked from commit `fa18f2d71e`)	2 years ago
Sean Bright	a7f1dbc462	core: Cleanup gerrit and JIRA references. (#40 ) * 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 (cherry picked from commit `6f218514fb`)	2 years ago
George Joseph	4d62b4c988	.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. (cherry picked from commit `49abb48678`)	2 years ago
George Joseph	54d872a10f	.github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS (cherry picked from commit `f508ae9c63`)	2 years ago
George Joseph	819e642c31	.github: Remove separate set labels step from new PR (cherry picked from commit `135764959e`)	2 years ago
George Joseph	06867f0705	.github: Refactor CP progress and add new PR test progress (cherry picked from commit `0395ee4760`)	2 years ago
Maximilian Fridrich	7140faf410	res_pjsip: mediasec: Add Security-Client headers after 401 When using mediasec, requests sent after a 401 must still contain the Security-Client header according to draft-dawes-sipcore-mediasec-parameter. Resolves: #48 (cherry picked from commit `ecd5f91125`)	2 years ago
George Joseph	ae9f5ca7ee	.github: Add cherry-pick test progress labels (cherry picked from commit `eece5824aa`)	2 years ago
Joshua C. Colp	db8c8627b5	LICENSE: Update link to trademark policy. Resolves: #43 (cherry picked from commit `f85587815f`)	2 years ago
InterLinked1	5dbd625bd6	chan_dahdi: Add dialmode option for FXS lines. (#36 ) 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. (cherry picked from commit `5d1dd11143`)	2 years ago
George Joseph	9a177ddacc	.github: Update issue templates (cherry picked from commit `a8a0adcdda`)	2 years ago
George Joseph	94a1f58a57	.github: Remove unnecessary parameter in CherryPickTest (cherry picked from commit `f4a3012619`)	2 years ago
George Joseph	48411942d6	Initial GitHub PRs (cherry picked from commit `ac403ec484`)	2 years ago
George Joseph	8f771ed7a4	Initial GitHub Issue Templates (cherry picked from commit `790f208c6e`)	2 years ago
Joshua C. Colp	09b313d606	pbx_dundi: Fix PJSIP endpoint configuration check. ASTERISK-28233 Change-Id: I0f11c096b307a6178e22ca49d9c756343f0e1fdc (cherry picked from commit `f5c9f7d3c8`)	2 years ago
Joshua Colp	6c15249f1d	Revert "app_queue: periodic announcement configurable start time." This reverts commit `c405630810`. Reason for revert: Causes segmentation fault. Change-Id: Ib0c8c592f8d4f0a5e3889aeadfe8bdcde800ba42 (cherry picked from commit `a8ce41b7f7`)	2 years ago
Naveen Albert	68a1c9b313	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 (cherry picked from commit `4e602a1afe`)	2 years ago
Naveen Albert	77384b93a1	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 (cherry picked from commit `1bbcb98558`)	2 years ago
The_Blode	0e7852c427	install_prereq: Add Linux Mint support. ASTERISK-30359 #close Change-Id: I9c140c7f12ca7dafe65c317f2a26122cf2c72556 (cherry picked from commit `e234531f5a`)	2 years ago
Henning Westerholt	e645ab6be9	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 (cherry picked from commit `91c4d78914`)	2 years ago
Naveen Albert	c9576cfeec	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 (cherry picked from commit `f5289a5483`)	2 years ago
Naveen Albert	0e02fd15cb	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 (cherry picked from commit `1ed147ffc1`)	2 years ago
George Joseph	363a569d99	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 (cherry picked from commit `a346fa54a5`)	2 years ago
Sean Bright	a42f692861	loader.c: Minor module key check simplification. Change-Id: I65aefd4434a783096165c179b5f94f2e4810dffe (cherry picked from commit `eb49555387`)	2 years ago
Sean Bright	aba5d71743	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 (cherry picked from commit `ee4278959a`)	2 years ago
Jaco Kroon	5aa0b0f961	res_calendar: output busy state as part of show calendar. Change-Id: I894e4ecc3e93db4ff7783d46266ba3c5e6ccda10 Signed-off-by: Jaco Kroon <jaco@uls.co.za> (cherry picked from commit `3ad6140b16`)	2 years ago
Mike Bradeen	815cbba409	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 (cherry picked from commit `1ad52a1263`)	2 years ago
Mike Bradeen	03da795e3e	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 (cherry picked from commit `b2e9419961`)	2 years ago
Mike Bradeen	33a9c3be91	format_sln: add .slin as supported file extension Adds '.slin' to existing supported file extensions: .sln and .raw ASTERISK-30465 Change-Id: Ice848addc03a64c8404b87cb5d3b13399c57e496 (cherry picked from commit `71a83c128d`)	2 years ago
Jaco Kroon	2383e8a598	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> (cherry picked from commit `c405630810`)	2 years ago
Naveen Albert	ea73a1f10d	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 (cherry picked from commit `595fe0f1cc`)	2 years ago
Naveen Albert	6d5998ae5c	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 (cherry picked from commit `48e3da5cdf`)	2 years ago
George Joseph	c71f73b778	make_version: Strip svn stuff and suppress ref HEAD errors * All of the code that used subversion has been removed. * When Asterisk is checked out from a tag or commit instead of one of the regular branches, git would emit messages like "fatal: ref HEAD is not a symbolic ref" which weren't fatal at all. Those are now suppressed. Change-Id: I2a11bc9ebbaf6dfa50f53516ede50a6bac65ca3c (cherry picked from commit `f41183e101`)	2 years ago
Fabrice Fontaine	7a6c146c4e	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 (cherry picked from commit `a10b4f5d2c`)	2 years ago
Mike Bradeen	20b0cfca39	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 (cherry picked from commit `f102e81d0f`)	2 years ago
Sean Bright	ec3add3c44	res_agi: RECORD FILE plays 2 beeps. Sending the "RECORD FILE" command without the optional `offset_samples` argument can result in two beeps playing on the channel. This bug has been present since Asterisk 0.3.0 (2003-02-06). ASTERISK-30457 #close Change-Id: I95e88aa59378784d7f0eb648843f090e6723b787 (cherry picked from commit `bcf0ffd278`)	2 years ago
Naveen Albert	44de51b6b0	app_senddtmf: Add SendFlash AMI action. Adds an AMI action to send a flash event on a channel. ASTERISK-30440 #close Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9 (cherry picked from commit `0c8b42ede3`)	2 years ago
Sean Bright	2268895b88	contrib: rc.archlinux.asterisk uses invalid redirect. `rc.archlinux.asterisk`, which explicitly requests bash in its shebang, uses the following command syntax: ${DAEMON} -rx "core stop now" > /dev/null 2&>1 The intent of which is to execute: ${DAEMON} -rx "core stop now" While sending both stdout and stderr to `/dev/null`. Unfortunately, because the `&` is in the wrong place, bash is interpreting the `2` as just an additional argument to the `$DAEMON` command and not as a file descriptor and proceeds to use the bashism `&>` to send stderr and stdout to a file named `1`. So we clean it up and just use bash's shortcut syntax. Issue raised and a fix suggested (but not used) by peutch on GitHub¹. ASTERISK-30449 #close 1. https://github.com/asterisk/asterisk/pull/31 Change-Id: Ie279bf4efb4d95cbf507313483d316e977303d19 (cherry picked from commit `5283d444d9`)	2 years ago
Fabrice Fontaine	86c9ee22f8	main/iostream.c: fix build with libressl Fix the following build failure with libressl by using SSL_is_server which is available since version 2.7.0 and `d7ec516916`: iostream.c: In function 'ast_iostream_close': iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'} 559 \| if (!stream->ssl->server) { \| ^~ ASTERISK-30107 #close Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9 Change-Id: Iea7f34970297f2fb50285d73462d0174ba7e9587 (cherry picked from commit `f34d9d5618`)	2 years ago

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