asterisk

Commit Graph

Author	SHA1	Message	Date
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
Sean Bright	76a2b2703f	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	2 years ago
Naveen Albert	cec0c5cd39	app_senddtmf: Add SendFlash AMI action. Adds an AMI action to send a flash event on a channel. ASTERISK-30440 #close Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9	2 years ago
Boris P. Korzun	e971396546	http.c: Minor simplification to HTTP status output. Change the HTTP status page (located at /httpstatus by default) by: * Combining the address and port into a single line. * Changing "SSL" to "TLS" ASTERISK-30433 #close Change-Id: Id2ccb1218f00a68424aca2b651647d8b1f549bcb	2 years ago
George Joseph	d433a048f7	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	2 years ago
Holger Hans Peter Freyther	92849c8c62	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	2 years ago
Sean Bright	cb8e3aa590	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	2 years ago
Fabrice Fontaine	b0635a5fcf	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	2 years ago
George Joseph	25f7753f46	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	2e78d1083e	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
Naveen Albert	73103bdcd5	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
Sean Bright	2e95f4713a	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
Nick French	a71f86d1a1	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
cmaj	fd95998b56	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	18a9540520	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
Mike Bradeen	152239b978	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	cee3564b45	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
Mike Bradeen	b44ffd0565	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
Sean Bright	f9a27f028f	app_queue: Minor docs and logging fixes for UnpauseQueueMember. ASTERISK-30417 #close Change-Id: I7534e7a925bf92a7b5a5347f5f54225768c162fe	2 years ago
Sean Bright	632da7c11a	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
Mike Bradeen	62a64686e2	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	2 years ago
Sean Bright	c85fc1278f	doxygen: Fix doxygen errors. Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	2 years ago
Naveen Albert	78f5a41d64	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	2 years ago
Mike Bradeen	2a066b6c2f	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	2 years ago
Naveen Albert	055e0a1571	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	2 years ago
Naveen Albert	d1bec3623e	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	2 years ago
sungtae kim	1da489a434	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	2 years ago
Sean Bright	c448dcd2f0	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	2 years ago
Mike Bradeen	6b03d60c7d	res_monitor: Remove deprecated module. ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50	2 years ago
Sean Bright	cbaba132a7	app_playback.c: Fix PLAYBACKSTATUS regression. In Asterisk 11, if a channel was redirected away during Playback(), the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 (specifically commit `7d9871b394`) that behavior was inadvertently changed and the same operation would result in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 behavior has been restored. Partial fix for ASTERISK~25661. Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff	2 years ago
George Joseph	91415a83d1	res_rtp_asterisk: Don't use double math to generate timestamps Rounding issues with double math were causing rtp timestamp slips in outgoing packets. We're now back to integer math and are getting no more slips. ASTERISK-30391 Change-Id: I6ba992b49ffdf9ebea074581dfa784a188c661a4	2 years ago
Mike Bradeen	e8f548c155	app_macro: Remove deprecated module. For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. Additionally, the following modules are impacted: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs a re-write app_queue - can no longer a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority ASTERISK-30304 Change-Id: I830daab293117179b8d61bd4df0d971a1b3d07f6	2 years ago
Alexei Gradinari	6ecec51e6a	format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...) Each playback of WAV files results in logging "Skipping unknown block 'LIST'". To prevent unnecessary flooding of this DEBUG log this patch replaces ast_log(LOG_DEBUG, ...) by ast_debug(1, ...). Change-Id: Iaa09cf19c5348a05385518fdb8cb181b45fe05f0	2 years ago
Igor Goncharovsky	410150235a	res_pjsip_rfc3326: Add SIP causes support for RFC3326 Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850). ASTERISK-30319 #close Change-Id: I3f55622dc680ce713a2ffb5a458ef5dd39fcf645	2 years ago
George Joseph	7dc8773178	res_rtp_asterisk: Asterisk Media Experience Score (MES) ----------------- This commit reinstates MES with some casting fixes to the functions in time.h that convert between doubles and timeval structures. The casting issues were causing incorrect timestamps to be calculated which caused transcoding from/to G722 to produce bad or no audio. ASTERISK-30391 ----------------- This module has been updated to provide additional quality statistics in the form of an Asterisk Media Experience Score. The score is avilable using the same mechanisms you'd use to retrieve jitter, loss, and rtt statistics. For more information about the score and how to retrieve it, see https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score * Updated chan_pjsip to set quality channel variables when a call ends. * Updated channels/pjsip/dialplan_functions.c to add the ability to retrieve the MES along with the existing rtcp stats when using the CHANNEL dialplan function. * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed checks for debugging purposes. * Added several function to time.h for manipulating time-in-samples and times represented as double seconds. * Updated rtp_engine.c to pass through the MES when stats are requested. Also debug output that dumps the stats when an rtp instance is destroyed. * Updated res_rtp_asterisk.c to implement the calculation of the MES. In the process, also had to update the calculation of jitter. Many debugging statements were also changed to be more informative. * Added a unit test for internal testing. The test should not be run during normal operation and is disabled by default. Change-Id: I4fce265965e68c3fdfeca55e614371ee69c65038	2 years ago
George Joseph	3a3d6c7dcb	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" This reverts commit `e66c5da145`. Reason for revert: Issue when transcoding to/from g722 Change-Id: I12853c5b1d3a77f5b9200f41908fd238a17159dc	2 years ago
Boris P. Korzun	8c9b37a539	http.c: Fix NULL pointer dereference bug If native HTTP is disabled but HTTPS is enabled and status page enabled too, Core/HTTP crashes while loading. 'global_http_server' references to NULL, but the status page tries to dereference it. The patch adds a check for HTTP is enabled. ASTERISK-30379 #close Change-Id: I11b02fc920b72aaed9c809fc43210523ccfdc249	2 years ago
Naveen Albert	c209064d66	loader: Allow declined modules to be unloaded. Currently, if a module declines to load, dlopen is called to register the module but dlclose never gets called. Furthermore, loader.c currently doesn't allow dlclose to ever get called on the module, since it declined to load and the unload function bails early in this case. This can be problematic if a module is updated, since the new module cannot be loaded into memory since we haven't closed all references to it. To fix this, we now allow modules to be unloaded, even if they never "loaded" in Asterisk itself, so that dlclose is called and the module can be properly cleaned up, allowing the updated module to be loaded from scratch next time. ASTERISK-30345 #close Change-Id: Ifc743aadfa85ebe3284e02a63e124dafa64988d5	2 years ago
Naveen Albert	f7726430b2	app_broadcast: Add Broadcast application Adds a new application, Broadcast, which can be used for one-to-many transmission and many-to-one reception of channel audio in Asterisk. This is similar to ChanSpy, except it is designed for multiple channel targets instead of a single one. This can make certain kinds of audio manipulation more efficient and streamlined. New kinds of audio injection impossible with ChanSpy are also made possible. ASTERISK-30180 #close Change-Id: I7ba72f765dbab9b58deeae028baca3f4f8377726	2 years ago

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