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Author	SHA1	Message	Date
George Joseph	790f208c6e	Initial GitHub Issue Templates	3 years ago
Joshua C. Colp	f5c9f7d3c8	pbx_dundi: Fix PJSIP endpoint configuration check. ... ASTERISK-28233 Change-Id: I0f11c096b307a6178e22ca49d9c756343f0e1fdc	3 years ago
Joshua Colp	a8ce41b7f7	Revert "app_queue: periodic announcement configurable start time." ... This reverts commit c405630810. Reason for revert: Causes segmentation fault. Change-Id: Ib0c8c592f8d4f0a5e3889aeadfe8bdcde800ba42	3 years ago
Naveen Albert	4e602a1afe	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	3 years ago
Naveen Albert	1bbcb98558	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	3 years ago
The_Blode	e234531f5a	install_prereq: Add Linux Mint support. ... ASTERISK-30359 #close Change-Id: I9c140c7f12ca7dafe65c317f2a26122cf2c72556	3 years ago
Henning Westerholt	91c4d78914	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	3 years ago
Naveen Albert	f5289a5483	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	3 years ago
Naveen Albert	1ed147ffc1	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	3 years ago
George Joseph	a346fa54a5	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	3 years ago
Mike Bradeen	c95f8ce69f	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	3 years ago
Sean Bright	eb49555387	loader.c: Minor module key check simplification. ... Change-Id: I65aefd4434a783096165c179b5f94f2e4810dffe	3 years ago
Sean Bright	ee4278959a	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	3 years ago
Jaco Kroon	3ad6140b16	res_calendar: output busy state as part of show calendar. ... Change-Id: I894e4ecc3e93db4ff7783d46266ba3c5e6ccda10 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	3 years ago
Sean Bright	f4469b09ca	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	3 years ago
Mike Bradeen	1ad52a1263	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	3 years ago
Mike Bradeen	b2e9419961	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	3 years ago
Mike Bradeen	71a83c128d	format_sln: add .slin as supported file extension ... Adds '.slin' to existing supported file extensions: .sln and .raw ASTERISK-30465 Change-Id: Ice848addc03a64c8404b87cb5d3b13399c57e496	3 years ago
Jaco Kroon	c405630810	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>	3 years ago
Naveen Albert	595fe0f1cc	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	3 years ago
Naveen Albert	48e3da5cdf	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	3 years ago
George Joseph	f41183e101	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	3 years ago
Fabrice Fontaine	a10b4f5d2c	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	3 years ago
Mike Bradeen	f102e81d0f	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	3 years ago
Sean Bright	bcf0ffd278	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	3 years ago
Naveen Albert	0c8b42ede3	app_senddtmf: Add SendFlash AMI action. ... Adds an AMI action to send a flash event on a channel. ASTERISK-30440 #close Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9	3 years ago
Sean Bright	5283d444d9	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	3 years ago
Fabrice Fontaine	f34d9d5618	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	3 years ago
Holger Hans Peter Freyther	2048877889	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	3 years ago
Asterisk Development Team	69297b59df	Update CHANGES and UPGRADE.txt for 18.17.0	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 years ago
Sean Bright	603675f285	app_queue: Minor docs and logging fixes for UnpauseQueueMember. ... ASTERISK-30417 #close Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe	3 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	3 years ago
Mike Bradeen	24e27a3c6e	res_pjsip: Upgraded bundled pjsip to 2.13 ... Removed multiple patches. Code chages in res_pjsip_pubsub due to changes in evsub. Pjsip now calls on_evsub_state() before on_rx_refresh(), so the sub tree deletion that used to take place in on_evsub_state() now must take place in on_rx_refresh(). Additionally, pjsip now requires that you send the NOTIFY from within on_rx_refresh(), otherwise it will assert when going to send the 200 OK. The idea is that it will look for this NOTIFY and cache it until after sending the response in order to deal with the self-imposed message mis-order. Asterisk previously dealt with this by pushing the NOTIFY in on_rx_refresh(), but pjsip now forces us to use it's method. Changes were required to configure in order to detect which way pjsip handles this as the two are not compatible for the reasons mentioned above. A corresponding change in testsuite is required in order to deal with the small interal timing changes caused by moving the NOTIFY send. ASTERISK-30325 Change-Id: I50b00cac89d950d3511d7b250a1c641965d9fe7f	3 years ago
Naveen Albert	d53c8fc4dc	app_signal: Add signaling applications ... Adds the Signal and WaitForSignal applications, which can be used for inter-channel signaling in the dialplan. Signal supports sending a signal to other channels listening for a signal of the same name, with an optional data payload. The signal is received by all channels waiting for that named signal. ASTERISK-29810 #close Change-Id: Ic34439de3d60f8609357666a465c354d81f5fef3	3 years ago
Sean Bright	41d3a57627	doxygen: Fix doxygen errors. ... Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	3 years ago
Mike Bradeen	ef6901e137	app_directory: add ability to specify configuration file ... Adds option to app_directory to specify a filename from which to read configuration instead of voicemail.conf ie; same => n,Directory(,,c(directory.conf)) This configuration should contain a list of extensions using the voicemail.conf format, ie; 2020=2020,Dog Dog,,,,attach=no|saycid=no|envelope=no|delete=no ASTERISK-30404 Change-Id: Id58ccb1344ad1e563fa10db12f172fbd104a9d13	3 years ago
Sean Bright	f67258d172	pbx_ael: Global variables are not expanded. ... Variable references within global variable assignments are now expanded rather than being included literally. ASTERISK-30406 #close Change-Id: I136e8d6395e90a4c92d9777a46a7bc3edb08d05d	3 years ago
Naveen Albert	7b9ef96173	func_json: Enhance parsing capabilities of JSON_DECODE ... Adds support for arrays to JSON_DECODE by allowing the user to print out entire arrays or index a particular key or print the number of keys in a JSON array. Additionally, adds support for recursively iterating a JSON tree in a single function call, making it easier to parse JSON results with multiple levels. A maximum depth is imposed to prevent potentially blowing the stack. Also fixes a bug with the unit tests causing an empty string to be printed instead of the actual test result. ASTERISK-29913 #close Change-Id: I603940b216a3911b498fc6583b18934011ef5d5b	3 years ago
Naveen Albert	ec0ca7dcbc	res_pjsip_session: Add overlap_context option. ... Adds the overlap_context option, which can be used to explicitly specify a context to use for overlap dialing extension matches, rather than forcibly using the context configured for the endpoint. ASTERISK-30262 #close Change-Id: Ibbcd4a8b11402428a187fb56b8d4e7408774a0db	3 years ago
sungtae kim	2f36ab7978	res_stasis_snoop: Fix snoop crash ... Added NULL pointer check and channel lock to prevent resource release while the chanspy is processing. ASTERISK-29604 Change-Id: Ibdc675f98052da32333b19685b1708a3751b6d24	3 years ago