asterisk

Commit Graph

Author	SHA1	Message	Date
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
Holger Hans Peter Freyther	8f628b13c1	res_http_media_cache: Introduce options and customize Make the existing CURL parameters configurable and allow to specify the usable protocols, proxy and DNS timeout. ASTERISK-30340 Change-Id: I2eb02ef44190e026716720419bcbdbcc8125777b (cherry picked from commit `2048877889`)	2 years ago
Asterisk Development Team	52f5afc63f	Update for 18.17.1	2 years ago
Mike Bradeen	2227bfbb2b	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
Sean Bright	c740de44f2	Revert "pbx_ael: Global variables are not expanded." This reverts commit `f67258d172`. Reason for revert: Behavior change that breaks existing dialplan. ASTERISK-30472 #close Change-Id: Ia03fa7f049f964ed396f6bf6d7ca096b665aa6b3	2 years ago
Asterisk Development Team	b03ffee26f	Update for 18.17.0	2 years ago
Asterisk Development Team	7a93d16d0f	Update for 18.17.0-rc1	2 years ago
Asterisk Development Team	69297b59df	Update CHANGES and UPGRADE.txt for 18.17.0	2 years ago
George Joseph	1ddfb7551a	res_pjsip: Replace invalid UTF-8 sequences in callerid name * Added a new function ast_utf8_replace_invalid_chars() to utf8.c that copies a string replacing any invalid UTF-8 sequences with the Unicode specified U+FFFD replacement character. For example: "abc\xffdef" becomes "abc\uFFFDdef". Any UTF-8 compliant implementation will show that character as a � character. * Updated res_pjsip:set_id_from_hdr() to use ast_utf8_replace_invalid_chars and print a warning if any invalid sequences were found during the copy. * Updated stasis_channels:ast_channel_publish_varset to use ast_utf8_replace_invalid_chars and print a warning if any invalid sequences were found during the copy. ASTERISK-27830 Change-Id: I4ffbdb19c80bf0efc675d40078a3ca4f85c567d8	2 years ago
Sean Bright	a724298da9	test.c: Avoid passing -1 to FD_* family of functions. This avoids buffer overflow errors when running tests that capture output from child processes. This also corrects a copypasta in an off-nominal error message. Change-Id: Ib482847a3515364f14c7e7a0c0a4213851ddb10d	2 years ago
Sean Bright	f4965b9430	test_crypto.c: Fix getcwd(…) build error. `getcwd(…)` is decorated with the `warn_unused_result` attribute and therefore needs its return value checked. Change-Id: Idcccb20a0abf293202c28633d0e9ee0f6a9dbe93	2 years ago
Naveen Albert	6aa28346cf	chan_iax2: Fix jitterbuffer regression prior to receiving audio. ASTERISK_29392 (a security fix) introduced a regression by not processing frames when we don't have an audio format. Currently, chan_iax2 only calls jb_get to read frames from the jitterbuffer when the voiceformat has been set on the pvt. However, this only happens when we receive a voice frame, which means that prior to receiving voice frames, other types of frames get stalled completely in the jitterbuffer. To fix this, we now fallback to using the format negotiated during call setup until we've actually received a voice frame with a format. This ensures we're always able to read from the jitterbuffer. ASTERISK-30354 #close ASTERISK-30162 #close Change-Id: Ie4fd1e8e088a145ad89e0427c2100a530e964fe9	2 years ago
Nick French	7196eb524b	pjproject_bundled: Fix cross-compilation with SSL libs. Asterisk makefiles auto-detect SSL library availability, then they assume that pjproject makefiles will also autodetect an SSL library at the same time, so they do not pass on the autodetection result to pjproject. This normally works, except the pjproject makefiles disables autodetection when cross-compiling. Fix by explicitly configuring pjproject to use SSL if we have been told to use it or it was autodetected ASTERISK-30424 #close Change-Id: I8fe2999ea46710e21d1d55a1bed92769c6ebded9	2 years ago
Mike Bradeen	fc5b7ab459	app_read: Add an option to return terminator on empty digits. Adds 'e' option to allow Read() to return the terminator as the dialed digits in the case where only the terminator is entered. ie; if "#" is entered, return "#" if the 'e' option is set and "" if it is not. ASTERISK-30411 Change-Id: I49f3221824330a193a20c660f99da0f1fc2cbbc5	2 years ago
cmaj	66f6d9113e	res_phoneprov.c: Multihomed SERVER cache prevention Phones moving between subnets on multi-homed server have their initially connected interface IP cached in the SERVER variable, even when it is not specified in the configuration files. This prevents phones from obtaining the correct SERVER variable value when they move to another subnet. ASTERISK-30388 #close Reported-by: cmaj Change-Id: I1d18987a9d58e85556b4c4a6814ce7006524cc92	2 years ago
Mike Bradeen	3b75e6d45e	app_directory: Add a 'skip call' option. Adds 's' option to skip calling the extension and instead set the extension as DIRECTORY_EXTEN channel variable. ASTERISK-30405 Change-Id: Ib9d9db1ba5b7524594c640461b4aa8f752db8299	2 years ago
Mike Bradeen	114630279d	res_pjsip: Prevent SEGV in pjsip_evsub_send_request contributed pjproject - patch to check sub->pending_notify in evsub.c:on_tsx_state before calling pjsip_evsub_send_request() res_pjsip_pubsub - change post pjsip 2.13 behavior to use pubsub_on_refresh_timeout to avoid the ao2_cleanup call on the sub_tree. This is is because the final NOTIFY send is no longer the last place the sub_tree is referenced. ASTERISK-30419 Change-Id: Ib5cc662ce578e9adcda312e16c58a10b6453e438	2 years ago
Mike Bradeen	f015d3e0cc	app_senddtmf: Add option to answer target channel. Adds a new option to SendDTMF() which will answer the specified channel if it is not already up. If no channel is specified, the current channel will be answered instead. ASTERISK-30422 Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085	2 years ago
Sean Bright	603675f285	app_queue: Minor docs and logging fixes for UnpauseQueueMember. ASTERISK-30417 #close Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe	2 years ago
Sean Bright	7687c75cb6	app_queue: Reset all queue defaults before reload. Several queue fields were not being set to their default value during a reload. Additionally added some sample configuration options that were missing from queues.conf.sample. Change-Id: I3a88c7877af91752b1b46a0c087384f7eb9c47e4	2 years ago

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33560 Commits (db8c8627b5019b564fb018cbf75b91997338a866) All Branches Search

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