6773 Commits (master)

Author	SHA1	Message	Date
Naveen Albert	0119f3ad48	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
George Joseph	1245e3330b	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
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	3 years ago
Sean Bright	7dc73c8151	loader.c: Minor module key check simplification. ... Change-Id: I65aefd4434a783096165c179b5f94f2e4810dffe	3 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	3 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	3 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	3 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	3 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	3 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	3 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	3 years ago
Sean Bright	c85fc1278f	doxygen: Fix doxygen errors. ... Change-Id: Ic50e95b4fc10f74ab15416d908e8a87ee8ec2f85	3 years ago
Mike Bradeen	6b03d60c7d	res_monitor: Remove deprecated module. ... ASTERISK-30303 Change-Id: I0462caefb4f9544e2e2baa23c498858310b52d50	3 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	3 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	3 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	3 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	3 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	3 years ago
Naveen Albert	ed77b365ce	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	3 years ago
Mike Bradeen	4095a382da	chan_sip: Remove deprecated module. ... ASTERISK-30297 Change-Id: Ic700168c80b68879d9cee8bb07afe2712fb17996	3 years ago
George Joseph	e66c5da145	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	3 years ago
Naveen Albert	f0962d00ae	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	3 years ago
Naveen Albert	f86d2a211c	pbx_builtins: Remove deprecated and defunct functionality. ... This removes the ImportVar and SetAMAFlags applications which have been deprecated since Asterisk 12, but were never removed previously. Additionally, it removes remnants of defunct options that themselves were removed years ago. ASTERISK-30335 #close Change-Id: I749520c7b08d4c9d5eebbf640d4fbc81950eda8d	3 years ago
Peter Fern	ee170ab166	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	3 years ago
Michael Kuron	6b8d3cb89a	manager: AOC-S support for AOCMessage ... ASTERISK-21502 Change-Id: I051b778f8c862d3b4794d28f2f3d782316707b08	3 years ago
Mike Bradeen	89a7d30a97	chan_mgcp: Remove deprecated module. ... Also removes res_pktcops to avoid merge conflicts with ASTERISK~30301. ASTERISK-30299 Change-Id: I41a316d327646a197b6f112f7f637aceb5111b41	3 years ago
Naveen Albert	1da5eb3795	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	c59eb7e6d8	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 (cherry picked from commit 81f10e847e)	3 years ago
Naveen Albert	b1d21f7667	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	315eb551db	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
Naveen Albert	6baa420986	file.c: Don't emit warnings on winks. ... Adds an ignore case for wink since it should pass through with no warning. ASTERISK-30290 #close Change-Id: Ieb7e34daa717357ac5c93efb0059f6c2321f16ad	3 years ago
Naveen Albert	5e35862109	translate.c: Prefer better codecs upon translate ties. ... If multiple codecs are available for the same resource and the translation costs between multiple codecs are the same, ties are currently broken arbitrarily, which means a lower quality codec would be used. This forces Asterisk to explicitly use the higher quality codec, ceteris paribus. ASTERISK-29455 Change-Id: I4b7297e1baca7aac14fe4a3c7538e18e2dbe9fd6	3 years ago
Naveen Albert	80e9e77261	manager: Update ModuleCheck documentation. ... The ModuleCheck XML documentation falsely claims that the module's version number is returned. This has not been the case since 14, since the version number is not available anymore, but the documentation was not changed at the time. It is now updated to reflect this. ASTERISK-30285 #close Change-Id: Idde2d1205a11f2623fa1ddab192faa3dc4081e91	3 years ago
Naveen Albert	e0d243396f	tcptls: Prevent crash when freeing OpenSSL errors. ... write_openssl_error_to_log has been erroneously using ast_free instead of free, which will cause a crash when MALLOC_DEBUG is enabled since the memory was not allocated by Asterisk's memory manager. This changes it to use the actual free function directly to avoid this. ASTERISK-30278 #close Change-Id: Iac8b6468b718075809c45d8ad16b101af21a474d	3 years ago
Frederic LE FOLL	54c1fd9829	Dialing API: Cancel a running async thread, may not cancel all calls ... race condition: ast_dial_join() may not cancel outgoing call, if function is called just after called party answer and before application execution (bit is_running_app not yet set). This fix adds ast_softhangup() calls in addition to existing pthread_kill() when is_running_app is not set. ASTERISK-30258 Change-Id: Idbdd5c15122159661aa8e996a42d5800083131e4	3 years ago
Naveen Albert	0825d26ad9	say: Don't prepend ampersand erroneously. ... Some logic in say.c for determining if we need to also add an ampersand for file seperation was faulty, as non-successful files would increment the count, causing a leading ampersand to be added improperly. This is fixed, and a unit test that captures this regression is also added. ASTERISK-30248 #close Change-Id: I02c1d3a11d82fe4ea8b462070cbd1effb5834d2b	3 years ago
Mike Bradeen	bb44e3edca	audiohook: add directional awareness ... Add enum to allow setting optional direction. If set to only one direction, only feed matching-direction frames to the associated slin factory. This prevents mangling the transcoder on non-mixed frames when the READ and WRITE frames would have otherwise required it. Also removes the need to mute or discard the un-wanted frames as they are no longer added in the first place. res_stasis_snoop is changed to use this addition to set direction on audiohook based on spy direction. If no direction is set, the ast_audiohook_init will init this enum to BOTH which maintains existing functionality. ASTERISK-30252 Change-Id: If8716bad334562a5d812be4eeb2a92e4f3be28eb	3 years ago
Naveen Albert	2b930d7c3b	cdr: Allow bridging and dial state changes to be ignored. ... Allows bridging, parking, and dial messages to be globally ignored for all CDRs such that only a single CDR record is generated per channel. This is useful when CDRs should endure for the lifetime of an entire channel and bridging and dial updates in the dialplan should not result in multiple CDR records being created for the call. With the ignore bridging option, bridging changes have no impact on the channel's CDRs. With the ignore dial state option, multiple Dials and their outcomes have no impact on the channel's CDRs. The last disposition on the channel is preserved in the CDR, so the actual disposition of the call remains available. These two options can reduce the amount of "CDR hacks" that have hitherto been necessary to ensure that CDR was not "spoiled" by these messages if that was undesired, such as putting a dummy optimization-disabled local channel between the caller and the actual call and putting the CDR on the channel in the middle to ensure that CDR would persist for the entire call and properly record start, answer, and end times. Enabling these options is desirable when calls correspond to the entire lifetime of channels and the CDR should reflect that. Current default behavior remains unchanged. ASTERISK-30091 #close Change-Id: I393981af42732ec5ac3ff9266444abb453b7c832	3 years ago
Naveen Albert	7fdc6334dc	db: Fix incorrect DB tree count for AMI. ... The DBGetTree AMI action's ListItem previously always reported 1, regardless of the count. This is corrected to report the actual count. ASTERISK-30245 #close patches: gettreecount.diff submitted by Birger Harzenetter (license 5870) Change-Id: I46d8992710f1b8524426b1255f57d1ef4a4934d4	3 years ago
Naveen Albert	7cbf779f13	features: Add no answer option to Bridge. ... Adds the n "no answer" option to the Bridge application so that answer supervision can not automatically be provided when Bridge is executed. Additionally, a mechanism (dialplan variable) is added to prevent bridge targets (typically the target of a masquerade) from answering the channel when they enter the bridge. ASTERISK-30223 #close Change-Id: I76f73fcd8e403bcd18f2abb40c658f537ac1ba6d	3 years ago
Philip Prindeville	a15fffa57e	test: initialize capture structure before freeing ... ASTERISK-30232 #close Change-Id: I2603e2cef8f93f6b0a6ef39f7eac744251bb3902	3 years ago
Jaco Kroon	2043234cf4	manager: be more aggressive about purging http sessions. ... If we find that n_max (currently hard wired to 1) sessions were purged, schedule the next purge for 1ms into the future rather than 5000ms (as per current). This way we will purge up to 1000 sessions per second rather than 1 every 5 seconds. This mitigates a build-up of sessions should http sessions gets established faster than 1 per 5 seconds. Change-Id: I9820d39aa080109df44fe98c1325cafae48d54f5 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	3 years ago
George Joseph	71de0babcc	res_crypto: Memory issues and uninitialized variable errors ... ASTERISK-30235 Change-Id: Ia1e326e7b52cd06fd5e6c9009e3e63193c92f6cd	3 years ago
George Joseph	bd2fb077ac	res_geolocation: Fix segfault when there's an empty element ... Fixed a segfault caused by var_list_from_loc_info() encountering an empty location info element. Fixed an issue in ast_strsep() where a value with only whitespace wasn't being preserved. Fixed an issue in ast_variable_list_from_quoted_string() where an empty value was considered a failure. ASTERISK-30215 Reported by: Dan Cropp Change-Id: Ieca64e061a6d9298f0196c694b60d986ef82613a	3 years ago
Philip Prindeville	82405752f7	main/utils: allow checking for command in $PATH ... ASTERISK-30037 Change-Id: I4b6f7264c8c737c476c798d2352f3232b263bbdf	3 years ago
Philip Prindeville	3f04dd5c01	test: Add ability to capture child process output ... ASTERISK-30037 Change-Id: Icbf84ce05addb197a458361c35d784e460d8d6c2	3 years ago
Naveen Albert	063722a64b	lock.c: Add AMI event for deadlocks. ... Adds an AMI event to indicate that a deadlock has likely started, when Asterisk is compiled with DETECT_DEADLOCKS enabled. This can make it easier to perform automated deadlock detection and take appropriate action (such as doing a core dump). Unlike the deadlock warnings, the AMI event is emitted only once per deadlock. ASTERISK-30161 #close Change-Id: Ifc6ed3e390f8b4cff7f8077a50e4d7a5b54e42fb	3 years ago
Naveen Albert	51e2a3ae65	pbx_variables: Use const char if possible. ... Use const char for char arguments to pbx_substitute_variables_helper_full_location that can do so (context and exten). ASTERISK-30209 #close Change-Id: I001357177e9c3dca2b2b4eebc5650c1095b3da6f	3 years ago
Naveen Albert	cd0d60a64b	cli: Prevent assertions on startup from bad ao2 refs. ... If "core show channels" is run before startup has completed, it is possible for bad ao2 refs to occur because the system is not yet fully initialized. This will lead to an assertion failing. To prevent this, initialization of CLI builtins is moved to be later along in the main load sequence. Core CLI commands are loaded at the same time, but channel-related commands are loaded later on. ASTERISK-29846 #close Change-Id: If6b3cde802876bd738c1b4cf2683bea6ddc615b6	3 years ago
Naveen Albert	3a95cadc1f	features: Add transfer initiation options. ... Adds additional control options over the transfer feature functionality to give users more control in how the transfer feature sounds and works. First, the "transfer" sound that plays when a transfer is initiated can now be customized by the user in features.conf, just as with the other transfer sounds. Secondly, the user can now specify the transfer extension in advance by using the TRANSFER_EXTEN variable. If a valid extension is contained in this variable, the call will automatically be transferred to this destination. Otherwise, it will fall back to collecting the extension from the user as is always done now. ASTERISK-29899 #close Change-Id: Ibff309caa459a2b958706f2ed0ca393b1ef502e3	3 years ago