6792 Commits (f96d7ef7b50e1b0b3b51a0d1321b16a59acae91b)

Author	SHA1	Message	Date
Holger Hans Peter Freyther	7699af00e1	stasis: Update the snapshot after setting the redirect ... The previous commit added the caller_rdnis attribute. Make it avialble during a possible ChanngelHangupRequest. (cherry picked from commit 56733c73b4)	1 year ago
Holger Hans Peter Freyther	28f52d35f3	ari: Provide the caller ID RDNIS for the channels ... Provide the caller ID RDNIS when available. This will allow an application to follow the redirect. (cherry picked from commit 157389bc59)	1 year ago
Brad Smith	089ddaaaed	main/utils: Implement ast_get_tid() for OpenBSD ... Implement the ast_get_tid() function for OpenBSD. OpenBSD supports getting the TID via getthrid(). (cherry picked from commit e7943dd4d9)	1 year ago
Naveen Albert	b47a403b80	core_local: Fix local channel parsing with slashes. ... Currently, trying to call a Local channel with a slash in the extension will fail due to the parsing of characters after such a slash as being dial modifiers. Additionally, core_local is inconsistent and incomplete with its parsing of Local dial strings in that sometimes it uses the first slash and at other times it uses the last. For instance, something like DAHDI/5 or PJSIP/device is a perfectly usable extension in the dialplan, but Local channels in particular prevent these from being called. This creates inconsistent behavior for users, since using a slash in an extension is perfectly acceptable, and using a Goto to accomplish this works fine, but if specified through a Local channel, the parsing prevents this. This fixes this by explicitly parsing options from the last slash in the extension, rather than the first one, which doesn't cause an issue for extensions with slashes. ASTERISK-30013 #close Resolves: #248 (cherry picked from commit 2191a0d33f)	1 year ago
Bastian Triller	7a1208410a	func_json: Fix crashes for some types ... This commit fixes crashes in JSON_DECODE() for types null, true, false and real numbers. In addition it ensures that a path is not deeper than 32 levels. Also allow root object to be an array. Add unit tests for above cases. (cherry picked from commit e6d5b8d8cf)	1 year ago
Eduardo	7a2305a08b	codec_builtin: Use multiples of 20 for maximum_ms ... Some providers require a multiple of 20 for the maxptime or fail to complete calls, e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used. Resolves: #260 (cherry picked from commit 11d87713eb)	1 year ago
George Joseph	9afa54b3b3	lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS ... Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS. Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just causes the lock calls to loop over trylock in 200us intervals until the lock is obtained and spits out log messages if it takes more than 5 seconds. From a code perspective, the only reason they were tied together was for logging. So... The ifdefs in lock.c were refactored to allow DETECT_DEADLOCKS to be enabled without also enabling DEBUG_THREADS. Resolves: #321 UserNote: You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS. This removes a significant amount of overhead if you just want to detect possible deadlocks vs needing full lock tracing. (cherry picked from commit 7e2243f9e1)	1 year ago
George Joseph	97674e1d20	asterisk.c: Use the euid's home directory to read/write cli history ... The CLI .asterisk_history file is read from/written to the directory specified by the HOME environment variable. If the root user starts asterisk with the -U/-G options, or with runuser/rungroup set in asterisk.conf, the asterisk process is started as root but then it calls setuid/setgid to set the new user/group. This does NOT reset the HOME environment variable to the new user's home directory though so it's still left as "/root". In this case, the new user will almost certainly NOT have access to read from or write to the history file. * Added function process_histfile() which calls getpwuid(geteuid()) and uses pw->dir as the home directory instead of the HOME environment variable. * ast_el_read_default_histfile() and ast_el_write_default_histfile() have been modified to use the new process_histfile() function. Resolves: #337 (cherry picked from commit 07cf37531a)	1 year ago
Mike Bradeen	3b61583c20	cel: add publish user event helper ... Add a wrapper function around ast_cel_publish_event that packs event and extras into a blob before publishing Resolves:#330 (cherry picked from commit 248b92563c)	1 year ago
George Joseph	d5a285c8f2	file.c: Add ability to search custom dir for sounds ... To better co-exist with sounds files that may be managed by packages, custom sound files may now be placed in AST_DATA_DIR/sounds/custom instead of the standard AST_DATA_DIR/sounds/<lang> directory. If the new "sounds_search_custom_dir" option in asterisk.conf is set to "true", asterisk will search the custom directory for sounds files before searching the standard directory. For performance reasons, the "sounds_search_custom_dir" defaults to "false". Resolves: #315 UserNote: A new option "sounds_search_custom_dir" has been added to asterisk.conf that allows asterisk to search AST_DATA_DIR/sounds/custom for sounds files before searching the standard AST_DATA_DIR/sounds/<lang> directory. (cherry picked from commit 4493d2b2fc)	1 year ago
George Joseph	172602d30e	make_buildopts_h, et. al. Allow adding all cflags to buildopts.h ... The previous behavior of make_buildopts_h was to not add the non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS, REF_DEBUG, etc. to the buildopts.h file because "it caused ccache to invalidate files and extended compile times". They're only defined by passing them on the gcc command line with '-D' options. In practice, including them in the include file rarely causes any impact because the only time ccache cares is if you actually change an option so the hit occurrs only once after you change it. OK so why would we want to include them? Many IDEs follow the include files to resolve defines and if the options aren't in an include file, it can cause the IDE to mark blocks of "ifdeffed" code as unused when they're really not. So... * Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H which tells make_buildopts_h to include the non-ABI-breaking flags in buildopts.h as well as the ABI-breaking ones. The default is disabled to preserve current behavior. As before though, only the ABI-breaking flags appear in AST_BUILDOPTS and only those are used to calculate AST_BUILDOPT_SUM. A new AST_BUILDOPT_ALL define was created to capture all of the flags. * make_version_c was streamlined to use buildopts.h and also to create asterisk_build_opts_all[] and ast_get_build_opts_all(void) * "core show settings" now shows both AST_BUILDOPTS and AST_BUILDOPTS_ALL. UserNote: The "Build Options" entry in the "core show settings" CLI command has been renamed to "ABI related Build Options" and a new entry named "All Build Options" has been added that shows both breaking and non-breaking options. (cherry picked from commit 06da7b342e)	1 year ago
Ben Ford	d44f07244a	manager.c: Prevent path traversal with GetConfig. ... When using AMI GetConfig, it was possible to access files outside of the Asterisk configuration directory by using filenames with ".." and "./" even while live_dangerously was not enabled. This change resolves the full path and ensures we are still in the configuration directory before attempting to access the file.	1 year ago
Naveen Albert	2d1e4aed4f	pbx.c: Fix gcc 12 compiler warning. ... Resolves: #277 (cherry picked from commit 5077301de6)	2 years ago
Maximilian Fridrich	b2cd0583c9	main/refer.c: Fix double free in refer_data_destructor + potential leak ... Resolves: #267 (cherry picked from commit fcdffe0074)	2 years ago
Joshua C. Colp	592c85b8de	manager: Tolerate stasis messages with no channel snapshot. ... In some cases I have yet to determine some stasis messages may be created without a channel snapshot. This change adds some tolerance to this scenario, preventing a crash from occurring. (cherry picked from commit 07d798875d)	2 years ago
Maximilian Fridrich	68fc422247	core/ari/pjsip: Add refer mechanism ... This change adds support for refers that are not session based. It includes a refer implementation for the PJSIP technology which results in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be triggered using the new ARI endpoint `/endpoints/refer`. Resolves: #71 UserNote: There is a new ARI endpoint `/endpoints/refer` for referring an endpoint to some URI or endpoint. (cherry picked from commit 8824b845c4)	2 years ago
Joshua C. Colp	fcf7187cd6	audiohook: Unlock channel in mute if no audiohooks present. ... In the case where mute was called on a channel that had no audiohooks the code was not unlocking the channel, resulting in a deadlock. Resolves: #233 (cherry picked from commit 5ebe1b61bd)	2 years ago
Mike Bradeen	714950bd7c	utils: add lock timestamps for DEBUG_THREADS ... Adds last locked and unlocked timestamps as well as a counter for the number of times the lock has been attempted (vs locked/unlocked) to debug output printed using the DEBUG_THREADS option. Resolves: #110 (cherry picked from commit 3acdffc17e)	2 years ago
Jaco Kroon	dafa9917ff	tcptls: when disabling a server port, we should set the accept_fd to -1. ... If we don't set this to -1 if the structure can be potentially re-used later then it's possible that we'll issue a close() on an unrelated file descriptor, breaking asterisk in other interesting ways. I believe this to be an unlikely scenario, but it costs nothing to be safe. Signed-off-by: Jaco Kroon <jaco@uls.co.za> (cherry picked from commit 4a637d6d11)	2 years ago
George Joseph	ee09bbbb5f	build: Fix a few gcc 13 issues ... * gcc 13 is now catching when a function is declared as returning an enum but defined as returning an int or vice versa. Fixed a few in app.h, loader.c, stasis_message.c. * gcc 13 is also now (incorrectly) complaining of dangling pointers when assigning a pointer to a local char array to a char *. Had to change that to an ast_alloca. Resolves: #155 (cherry picked from commit 6c8b23a688)	2 years ago
Ben Ford	8da7b4606b	AMI: Add CoreShowChannelMap action. ... Adds a new AMI action (CoreShowChannelMap) that takes in a channel name and provides a list of all channels that are connected to that channel, following local channel connections as well. Resolves: #104 UserNote: New AMI action CoreShowChannelMap has been added. (cherry picked from commit e72b277828)	2 years ago
Mike Bradeen	26401ddcc0	indications: logging changes ... Increase verbosity to indicate failure due to missing country and to specify default on CLI dump Resolves: #89 (cherry picked from commit f19a6cf11e)	2 years ago
Naveen Albert	ba1c363d3e	callerid: Allow specifying timezone for date/time. ... The Caller ID generation routine currently is hardcoded to always use the system time zone. This makes it possible to optionally specify any TZ-format time zone. Resolves: #98 ASTERISK-30330 (cherry picked from commit d3227a614a)	2 years ago
Naveen Albert	ca51a25056	asterisk.c: Fix option warning for remote console. ... Commit 09e989f972 categorized the T option as not being compatible with remote consoles, but they do affect verbose messages with remote console. This fixes this. Resolves: #102 (cherry picked from commit cfae64a70d)	2 years ago
Sean Bright	0e4a9d7251	xml.c: Process XML Inclusions recursively. ... 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 (cherry picked from commit 19973b73fd)	2 years ago
Mike Bradeen	21ee5f705a	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 8d3d8fcdcd)	2 years ago
Naveen Albert	59c9e4531e	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 66a7cff57e)	2 years ago
George Joseph	d0ce6d4ee5	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 a0fd95ef52)	2 years ago
Sean Bright	255565e240	loader.c: Minor module key check simplification. ... Change-Id: I65aefd4434a783096165c179b5f94f2e4810dffe (cherry picked from commit 6dab013e49)	2 years ago
Sean Bright	6ff3258c17	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 34ff836db5)	2 years ago
Mike Bradeen	2ac8388c6d	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 ffe346b2de)	2 years ago
Mike Bradeen	e00eaa74e3	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 fa635a872e)	2 years ago
Mike Bradeen	5a85b372af	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 405211eff7)	2 years ago
Fabrice Fontaine	b17177c419	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 030b7b9009)	2 years ago
George Joseph	ceda5a9859	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	e5c5cd6e25	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	96d9ad51ac	doxygen: Fix doxygen errors. ... Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	2 years ago
George Joseph	4710f37ef6	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	62ca063fca	Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)" ... This reverts commit d454801c2d. Reason for revert: Issue when transcoding to/from g722 Change-Id: I09f49e171b1661548657a9ba7a978c29d0b5be86	2 years ago
Naveen Albert	d33bd6d67a	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	7b8f7428da	manager: Fix appending variables. ... The if statement here is always false after the for loop finishes, so variables are never appended. This removes that to properly append to the end of the variable list. ASTERISK-30351 #close Reported by: Sebastian Gutierrez Change-Id: I1b7f8b85a8918f6a814cb933a479d4278cf16199	2 years ago
Boris P. Korzun	edc90c96ac	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
George Joseph	d454801c2d	res_rtp_asterisk: Asterisk Media Experience Score (MES) ... 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. ASTERISK-30280 Change-Id: I458cb9a311e8e5dc1db769b8babbcf2e093f107a	2 years ago
Naveen Albert	cc8d9b947b	pbx_app: Update outdated pbx_exec channel snapshots. ... pbx_exec makes a channel snapshot before executing applications. This doesn't cause an issue during normal dialplan execution where pbx_exec is called over and over again in succession. However, if pbx_exec is called "one off", e.g. using ast_pbx_exec_application, then a channel snapshot never ends up getting made after the executed application returns, and inaccurate snapshot information will linger for a while, causing "core show channels", etc. to show erroneous info. This is fixed by manually making a channel snapshot at the end of ast_pbx_exec_application, since we anticipate that pbx_exec might not get called again immediately. ASTERISK-30367 #close Change-Id: I2a5131053aa9d11badbc0ef2ef40b1f83d0af086	2 years ago
Peter Fern	58404b5c22	streams: Ensure that stream is closed in ast_stream_and_wait on error ... When ast_stream_and_wait returns an error (for example, when attempting to stream to a channel after hangup) the stream is not closed, and callers typically do not check the return code. This results in leaking file descriptors, leading to resource exhaustion. This change ensures that the stream is closed in case of error. ASTERISK-30198 #close Reported-by: Julien Alie Change-Id: Ie46b67314590ad75154595a3d34d461060b2e803	2 years ago
Michael Kuron	5c114dcb4a	manager: AOC-S support for AOCMessage ... ASTERISK-21502 Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08	3 years ago
Naveen Albert	52c7d3ed07	xmldoc: Allow XML docs to be reloaded. ... The XML docs are currently only loaded on startup with no way to update them during runtime. This makes it impossible to load modules that use ACO/Sorcery (which require documentation) if they are added to the source tree and built while Asterisk is running (e.g. external modules). This adds a CLI command to reload the XML docs during runtime so that documentation can be updated without a full restart of Asterisk. ASTERISK-30289 #close Change-Id: I4f265b0e5517e757c5453a0f241201a5788d3a07	3 years ago
Mike Bradeen	81f10e847e	manager: prevent file access outside of config dir ... Add live_dangerously flag to manager and use this flag to determine if a configuation file outside of AST_CONFIG_DIR should be read. ASTERISK-30176 Change-Id: I46b26af4047433b49ae5c8a85cb8cda806a07404	3 years ago
Naveen Albert	c7df5ee7c1	pbx_builtins: Allow Answer to return immediately. ... The Answer application currently waits for up to 500ms for media, even if users specify a different timeout. This adds an option to not wait for media on the channel by doing a raw answer instead. The default 500ms threshold is also documented. ASTERISK-30308 #close Change-Id: Id59cd340c44b8b8b2384c479e17e5123e917cba4	3 years ago
Maximilian Fridrich	60b81eabe0	core & res_pjsip: Improve topology change handling. ... This PR contains two relatively separate changes in channel.c and res_pjsip_session.c which ensure that topology changes are not ignored in cases where they should be handled. For channel.c: The function ast_channel_request_stream_topology_change only triggers a stream topology request change indication, if the channel's topology does not equal the requested topology. However, a channel could be in a state where it is currently "negotiating" a new topology but hasn't updated it yet, so the topology request change would be lost. Channels need to be able to handle such situations internally and stream topology requests should therefore always be passed on. In the case of chan_pjsip for example, it queues a session refresh (re-INVITE) if it is currently in the middle of a transaction or has pending requests (among other reasons). Now, ast_channel_request_stream_topology_change always indicates a stream topology request change even if the requested topology equals the channel's topology. For res_pjsip_session.c: The function resolve_refresh_media_states does not process stream state changes if the delayed active state differs from the current active state. I.e. if the currently active stream state has changed between the time the sip session refresh request was queued and the time it is being processed, the session refresh is ignored. However, res_pjsip_session contains logic that ensures that session refreshes are queued and re-queued correctly if a session refresh is currently not possible. So this check is not necessary and led to some session refreshes being lost. Now, a session refresh is done even if the delayed active state differs from the current active state and it is checked whether the delayed pending state differs from the current active - because that means a refresh is necessary. Further, the unit test of resolve_refresh_media_states was adapted to reflect the new behavior. I.e. the changes to delayed pending are prioritized over the changes to current active because we want to preserve the original intention of the pending state. ASTERISK-30184 Change-Id: Icd0703295271089057717006730b555b9a1d4e5a	3 years ago