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Author	SHA1	Message	Date
Naveen Albert	51146db5fe	channel.c: Clean up debug level 1. ... Although there are 10 debugs levels, over time, many current debug calls have come to use inappropriately low debug levels. In particular, a select few debug calls (currently all debug 1) can result in thousands of debug messages per minute for a single call. This can adds a lot of noise to core debug which dilutes the value in having different debug levels in the first place, as these log messages are from the core internals are are better suited for higher debug levels. Some debugs levels are thus adjusted so that debug level 1 is not inappropriately overloaded with these extremely high-volume and general debug messages. ASTERISK-29897 #close Change-Id: I55a71598993552d3d64a401a35ee99474770d4b4	4 years ago
Naveen Albert	e16fce2251	configs, LICENSE: remove pbx.digium.com. ... pbx.digium.com no longer accepts IAX2 calls and there are no plans for it to come back. Accordingly, nonworking IAX2 URIs are removed from both the LICENSE file and the sample config. ASTERISK-29923 #close Change-Id: I257c54d4d812ed6b4bd4cbec2cd7ebe2b87b5bad	4 years ago
Naveen Albert	d5c72f35fd	asterisk: Add macro for curl user agent. ... Currently, each module that uses libcurl duplicates the standard Asterisk curl user agent. This adds a global macro for the Asterisk user agent used for curl requests to eliminate this duplication. ASTERISK-29861 #close Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d	4 years ago
Naveen Albert	eb77229b53	documentation: Add since tag to xmldocs DTD ... Adds the since tag to the documentation DTD so that individual applications, functions, etc. can now specify when they were added to Asterisk. This tag is added at the individual application, function, etc. level as opposed to at the module level because modules can expand over time as new functionality is added, and granularity only to the module level would generally not be useful. This enables the ability to more easily determine when new functionality was added to Asterisk, down to minor version as opposed to just by major version. This makes it easier for users to write more portable dialplan if desired to not use functionality that may not be widely available yet. ASTERISK-29896 #close Change-Id: Ibbb35c702d8038bdc3fd0a944fbfa69384cc15d5	4 years ago
Naveen Albert	4cb4a3ed01	res_stir_shaken: refactor utility function ... Refactors temp file utility function into file.c. ASTERISK-29809 #close Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b	4 years ago
Naveen Albert	19481fc1f0	app_voicemail: Emit warning if asking for nonexistent mailbox. ... Currently, if VoiceMailMain is called with a mailbox, if that mailbox doesn't exist, then the application silently falls back to prompting the user for the mailbox, as if no arguments were provided. However, if a specific mailbox is requested and it doesn't exist, then no warning at all is emitted. This fixes this behavior to now warn if a specifically requested mailbox could not be accessed, before falling back to prompting the user for the correct mailbox. ASTERISK-29920 #close Change-Id: Ib4093b88cd661a2cabc5d685777d4e2f0ebd20a4	4 years ago
Alexei Gradinari	76f90bff75	res_pjsip_pubsub: fix Batched Notifications stop working ... If Subscription refresh occurred between when the batched notification was scheduled and the serialized notification was to be sent, then new schedule notification task would never be added. There are 2 threads: thread #1. ast_sip_subscription_notify is called, if notification_batch_interval then call schedule_notification. 1.1. The schedule_notification checks notify_sched_id > -1 not true, then send_scheduled_notify = 1 notify_sched_id = ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb.... 1.2. The sched_cb pushes task serialized_send_notify to serializer and returns 0 which means no reschedule. 1.3. The serialized_send_notify checks send_scheduled_notify if it's false the just returns. BUT notify_sched_id is still set, so no more ast_sched_add. thread #2. pubsub_on_rx_refresh is called 2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer 2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout which calls send_notify 2.3. The send_notify set send_scheduled_notify = 0; The serialized_send_notify should always unset notify_sched_id. ASTERISK-29904 #close Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875	4 years ago
Naveen Albert	257590b4e2	func_db: Add validity check for key names when writing. ... Adds a simple sanity check for key names when users are writing data to AstDB. This captures four cases indicating malformed keynames that generally result in bad data going into the DB that the user didn't intend: an empty key name, a key name beginning or ending with a slash, and a key name containing two slashes in a row. Generally, this is the result of a variable being used in the key name being empty. If a malformed key name is detected, a warning is emitted to indicate the bug in the dialplan. ASTERISK-29925 #close Change-Id: Ifc08a9fe532a519b1b80caca1aafed7611d573bf	4 years ago
Alexei Gradinari	e85ee05e9d	res_pjsip_pubsub: provide a display name for RLS subscriptions ... Whereas BLFs allow to show a display name for each RLS entry, the asterisk provides only the extension now. This is not end user friendly. This commit adds a new resource_list option, resource_display_name, to indicate whether display name of resource or the resource name being provided for RLS entries. If this option is enabled, the Display Name will be provided. This option is disabled by default to remain the previous behavior. If the 'event' set to 'presence' or 'dialog' the non-empty HINT name will be set as the Display Name. The 'message-summary' is not supported yet. ASTERISK-29891 #close Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681	4 years ago
Naveen Albert	57685c139b	documentation: Adds missing default attributes. ... The configObject tag contains a default attribute which allows the default value to be specified, if applicable. This allows for the default value to show up specially on the wiki in a way that is clear to users. There are a couple places in the tree where default values are included in the description as opposed to as attributes, which means these can't be parsed specially for the wiki. These are changed to use the attribute instead of being included in the text description. ASTERISK-29898 #close Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755	4 years ago
Naveen Albert	e8f8d84ffd	cli: Add core dump info to core show settings. ... Adds two pieces of information to the core show settings command which are useful in the context of getting backtraces. The first is to display whether or not Asterisk would generate a core dump if it were to crash. The second is to show the current running directory of Asterisk. ASTERISK-29866 #close Change-Id: Ic42c0a9ecc233381aad274d86c62808d1ebb4d83	4 years ago
Naveen Albert	b54060021f	app_mp3: Document and warn about HTTPS incompatibility. ... mpg123 doesn't support HTTPS, but the MP3Player application doesn't document this or warn the user about this. HTTPS streams have become more common nowadays and users could reasonably try to play them without being aware they should use the HTTP stream instead. This adds documentation to note this limitation. It also throws a warning if users try to use the HTTPS stream to tell them to use the HTTP stream instead. ASTERISK-29900 #close Change-Id: Ie3b029be5258c5a701f71ed3b1a7a80d1e03b827	4 years ago
Naveen Albert	ba2f780930	app_mf: Add max digits option to ReceiveMF. ... Adds an option to the ReceiveMF application to allow specifying a maximum number of digits. Originally, this capability was not added to ReceiveMF as it was with ReceiveSF because typically a ST digit is used to denote that sending of digits is complete. However, there are certain signaling protocols which simply transmit a digit (such as Expanded In-Band Signaling) and for these, it's necessary to be able to read a certain number of digits, as opposed to until receiving a ST digit. This capability is added as an option, as opposed to as a parameter, to remain compatible with existing usage (and not shift the parameters). ASTERISK-29877 #close Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b	4 years ago
Mike Bradeen	5764414d10	taskprocessor.c: Prevent crash on graceful shutdown ... When tps_shutdown is called as part of the cleanup process there is a chance that one of the taskprocessors that references the tps_singletons object is still running. The change is to allow for tps_shutdown to check tps_singleton's container count and give the running taskprocessors a chance to finish. If after AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT (10) seconds there are still container references we shutdown anyway as this is most likely a bug due to a taskprocessor not being unreferenced. ASTERISK-29365 Change-Id: Ia932fc003d316389b9c4fd15ad6594458c9727f1	4 years ago
Alexei Gradinari	5f22f586ff	app_queue: load queues and members from Realtime when needed ... There are a lot of Queue AMI actions and Queue applications which do not load queue and queue members from Realtime. AMI actions QueuePause - if queue not in memory - response "Interface not found". QueueStatus/QueueSummary - if queue not in memory - empty response. Applications: PauseQueueMember - if queue not in memory Attempt to pause interface %s, not found UnpauseQueueMember - if queue not in memory Attempt to unpause interface xxxxx, not found This patch adds a new function load_realtime_queues which loads queue and queue members for desired queue or all queues and all members if param 'queuename' is NULL or empty. Calls the function load_realtime_queues when needed. Also this patch fixes leak of ast_config in function set_member_value. Also this patch fixes incorrect LOG_WARNING when pausing/unpausing already paused/unpaused member. The function ast_update_realtime returns 0 when no record modified. So 0 is not an error to warn about. ASTERISK-29873 #close ASTERISK-18416 #close ASTERISK-27597 #close Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c	4 years ago
Sean Bright	67eb7c9e85	manager.c: Simplify AMI ModuleCheck handling ... This code was needlessly complex and would fail to properly delimit the response message if LOW_MEMORY was defined. Change-Id: Iae50bf09ef4bc34f9dc4b49435daa76f8b2c5b6e	4 years ago
Mark Petersen	97ac3c1385	res_prometheus.c: missing module dependency ... added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO which fixes issue with module crashes on load "FRACK!, Failed assertion" ASTERISK-29871 Change-Id: Ia0f49d048427a40e1b763296b834a52a03610096	4 years ago
Sean Bright	dfbb547f6a	res_pjsip.c: Correct minor typos in 'realm' documentation. ... Change-Id: I886936b808def5540d40071321e72f6bfa19063a	4 years ago
Sean Bright	b42dd930f4	manager.c: Generate valid XML if attribute names have leading digits. ... The XML Manager Event Interface (amxml) now generates attribute names that are compliant with the XML 1.1 specification. Previously, an attribute name that started with a digit would be rendered as-is, even though attribute names must not begin with a digit. We now prefix attribute names that start with a digit with an underscore ('_') to prevent XML validation failures. This is not backwards compatible but my assumption is that compliant XML parsers would already have been complaining about this. ASTERISK-29886 #close Change-Id: Icfaa56a131a082d803e9b7db5093806d455a0523	4 years ago
Asterisk Development Team	3d390c4df7	Update CHANGES and UPGRADE.txt for 19.2.0	4 years ago
Sean Bright	8b4531051b	build_tools/make_version: Fix bashism in comparison. ... In POSIX sh (which we indicate in the shebang), there is no == operator. Change-Id: Ic03d38214d14cdf329b0ba272279a815bb532965	4 years ago
George Joseph	3f108867d6	bundled_pjproject: Add additional multipart search utils ... Added the following APIs: pjsip_multipart_find_part_by_header() pjsip_multipart_find_part_by_header_str() pjsip_multipart_find_part_by_cid_str() pjsip_multipart_find_part_by_cid_uri() Change-Id: I6aee3dcf59eb171f93aae0f0564ff907262ef40d	4 years ago
Mark Petersen	25edc8ba47	chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN ... resolve issue with pickup on device that uses "183" and not "180" ASTERISK-29832 Change-Id: I4c7d223870f8ce9a7354e0f73d4e4cb2e8b58841	4 years ago
George Joseph	8bc6511c0c	build: Add "basebranch" to .gitreview ... If you have a development branch for a major project that will receive gerrit reviews it'll probably be named something like "development/16/newproject". That will necessitate setting "defaultbranch=development/16/newproject" in .gitreview. The make_version script uses that variable to construct the asterisk version however, which results in versions like "GIT-development/16/newproject-ee582a8c7b" which is probably not what you want. Worse, since the download_externals script uses make_version to construct the URL to download the binary codecs or DPMA. Since it's expecting a simple numeric version, the downloads will fail. To get this to work, a new variable "basebranch" has been added to .gitreview and make_version has been updated to use that instead of defaultversion: .gitreview: defaultbranch=development/16/myproject basebranch=16 Now git-review will send the reviews to the proper branch (development/16/myproject) but the version will still be constructed using the simple branch number (16). If "basebranch" is missing from .gitreview, make_version will fall back to using "defaultbranch". Change-Id: I2941a3b21e668febeb6cfbc1a7bb51a67726fcc4	4 years ago
George Joseph	4142e548e8	res_pjsip_outbound_authenticator_digest: Prevent ABRT on cleanup ... In dev mode, if you call pjsip_auth_clt_deinit() with an auth_sess that hasn't been initialized, it'll assert and abort. If digest_create_request_with_auth() fails to find the proper auth object however, it jumps to its cleanup which does exactly that. So now we no longer attempt to call pjsip_auth_clt_deinit() if we never actually initialized it. ASTERISK-29888 Change-Id: Ib6171c25c9fe8e61cc8d11129e324c021bc30b62	4 years ago
Naveen Albert	1a8d320a29	cdr: allow disabling CDR by default on new channels ... Adds a new option, defaultenabled, to the CDR core to control whether or not CDR is enabled on a newly created channel. This allows CDR to be disabled by default on new channels and require the user to explicitly enable CDR if desired. Existing behavior remains unchanged. ASTERISK-29808 #close Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1	4 years ago
Naveen Albert	cf17b3d211	func_frame_drop: Fix typo referencing wrong buffer ... In order to get around the issue of certain frames having names that could overlap, func_frame_drop surrounds names with commas for the purposes of comparison. The buffer is allocated and printed to properly, but the original buffer is used for comparison. In most cases, this wouldn't have had any effect, but that was not the intention behind the buffer. This updates the code to reference the modified buffer instead. ASTERISK-29854 #close Change-Id: I430b52e14e712d0e62a23aa3b5644fe958b684a7	4 years ago
Naveen Albert	d107ebd94a	res_tonedetect: Fixes some logic issues and typos ... Fixes some minor logic issues with the module: Previously, the OPT_END_FILTER flag was getting tested before options were parsed, so it could never evaluate to true (wrong ordering). Additionally, the initially parsed timeout (float) needs to be compared with 0, not the result int which is set afterwards (wrong variable). ASTERISK-29857 #close Change-Id: I0062bce3b391c15e5df7a714780eeaa96dd93d4c	4 years ago
Torrey Searle	34065c7d84	res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf ... When generating dtmfs, asterisk can incorrectly think packet loss occured during the dtmf generation, resulting in a jump in sequence numbers when forwarding voice frames resumes. This patch forces asterisk to re-learn the expected sequence number after each DTMF to avoid this ASTERISK-29869 #close Change-Id: Icc7de3d947b207b82c99d3c327af8095884df853	4 years ago
Sean Bright	8d1858e570	build: Rebuild configure and autoconfig.h.in ... autoconfigh.h.in was missed in the original review for this issue. Additionally it looks like I have newer pkg-config autoconf macros on my development machine. ASTERISK-29817 Change-Id: I3c85a4de82c5d7d6e0e23dad4c33bb650a86a57b	4 years ago
Kevin Harwell	9f5c1d30ed	res_http_websocket: Add a client connection timeout ... Previously there was no way to specify a connection timeout when attempting to connect a websocket client to a server. This patch makes it possible to now do such. Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d	4 years ago
Mike Bradeen	c1ca028a0b	sched: fix and test a double deref on delete of an executing call back ... sched: Avoid a double deref when AST_SCHED_DEL_UNREF is called on an executing call-back. This is done by adding a new variable 'rescheduled' to the struct sched which is set in ast_sched_runq and checked in ast_sched_del_nonrunning. ast_sched_del_nonrunning is a replacement for now deprecated ast_sched_del which returns a new possible value -2 if called on an executing call-back with rescheduled set. ast_sched_del is modified to call ast_sched_del_nonrunning to maintain existing code. AST_SCHED_DEL_UNREF is also updated to look for the -2 in which case it will not throw a warning or invoke refcall. test_sched: Add a new unit test sched_test_freebird that will check the reference count in the resolved scenario. ASTERISK-29698 Change-Id: Icfb16b3acbc29cf5b4cef74183f7531caaefe21d	5 years ago
Mark Petersen	3522b07b64	app_queue.c: Queue don't play "thank-you" when here is no hold time announcements ... if holdtime is (0 min, 0 sec) there is no hold time announcements we should then also not playing queue-thankyou ASTERISK-29831 Change-Id: Ic7e51dcde526b23f1cd8d24e1d1e2d81e10f9d2c	5 years ago
Luke Escude	c3d571f335	res_pjsip_sdp_rtp.c: Support keepalive for video streams. ... ASTERISK-28890 #close Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1	5 years ago
Michał Górny	43661122f5	build_tools/make_version: Fix sed(1) syntax compatibility with NetBSD ... Fix the sed(1) invocation used to process git-svn-id not to use "\s" that is a GNU-ism and is not supported by NetBSD sed. As a result, this call did not work properly and make_version did output the full git-svn-id line rather than the revision. ASTERISK-29852 Change-Id: Ie4b406e2748920643446851a0a252a4ca7245772	5 years ago
Michał Górny	de420bffa7	main/utils: Implement ast_get_tid() for NetBSD ... Implement the ast_get_tid() function for NetBSD system. NetBSD supports getting the TID via _lwp_self(). ASTERISK-29850 Change-Id: If57fd3f9ea15ef5d010bfbdcbbbae9b379f72f8c	5 years ago
Michał Górny	a7f7e5ab76	main: Enable rdtsc support on NetBSD ... Enable the Linux rdtsc implementation on NetBSD as well. The assembly works correctly there. ASTERISK-29851 Change-Id: I460ad9b4d971913420ecb84186f5ba5ab03f6f37	5 years ago
Michał Górny	addb7278d4	include: Remove unimplemented HMAC declarations ... Remove the HMAC declarations from the includes. They are not implemented nor used anywhere, and their presence breaks the build on NetBSD that delivers an incompatible hmac() function in <stdlib.h>. ASTERISK-29818 Change-Id: I0c4b88645e30174b1b63846a6b328625b69c2ea7	5 years ago
Naveen Albert	c452cc3f22	res_rtp_asterisk: Fix typo in flag test/set ... The code currently checks to see if an RFC3389 warning flag is set, except if it is, it merely sets the flag again, the logic of which doesn't make any sense. This adjusts the if comparison to check if the flag has NOT been set, and if so, emit a notice log event and set the flag so that future frames do not cause an event to be logged. ASTERISK-29856 #close Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1	5 years ago
Michał Górny	e32aec6c09	BuildSystem: Fix misdetection of gethostbyname_r() on NetBSD ... Fix the configure script not to detect the presence of gethostbyname_r() on NetBSD incorrectly. NetBSD includes it as an internal libc symbol that is not exposed in system headers and that is incompatible with other implementations. In order to avoid misdetecting it, perform the symbol check only if the declaration is found in the public header first. ASTERISK-29817 Change-Id: Iafa359b09908251bcd299ff54be003ea129b9eda	5 years ago
Naveen Albert	d3bd4a3a73	frame.h: Fix spelling typo ... Fixes CNG description from "noice" to "noise". ASTERISK-29855 #close Change-Id: Ie7cbbd7d72b426693df7447384ff8700318cd36d	5 years ago
George Joseph	3a18886803	bundled_pjproject: Fix srtp detection ... Reverted recent change that set '--with-external-srtp' instead of '--without-external-srtp'. Since Asterisk handles all SRTP, we don't need it enabled in pjproject at all. ASTERISK-29867 Change-Id: I2ce1bdd30abd21c062eac8f8fefe9b898787b801	5 years ago
George Joseph	eb1e20cf89	res_pjsip: Make message_filter and session multipart aware ... Neither pjsip_message_filter's filter_on_tx_message() nor res_pjsip_session's session_outgoing_nat_hook() were multipart aware and just assumed that an SDP would be the only thing in a message body. Both were changed to use the new pjsip_get_sdp_info() function which searches for an sdp in both single- and multi- part message bodies. ASTERISK-29813 Change-Id: I8f5b8cfdc27f1d4bd3e7491ea9090951a4525c56	5 years ago
George Joseph	f41cce9c12	build: Fix issues building pjproject ... The change to allow easier hacking on bundled pjproject created a few issues: * The new Makefile was trying to run the bundled make even if PJPROJECT_BUNDLED=no. third-party/Makefile now checks for PJPROJECT_BUNDLED and JANSSON_BUNDLED and skips them if they are "no". * When building with bundled, config_site.h was being copied only if a full make or a "make main" was done. A "make res" would fail all the pjsip modules because they couldn't find config_site.h. The Makefile now copies config_site.h and asterisk_malloc_debug.h into the pjproject source tree when it's "configure" is performed. This is how it used to be before the big change. ASTERISK-29858 Change-Id: I9427264fa3cb8b3f59a95e5f9693eac236a6f76d	5 years ago
George Joseph	a611c07dd4	res_pjsip: Add utils for checking media types ... Added two new functions to assist checking media types... * ast_sip_are_media_types_equal compares two pjsip_media_types. * ast_sip_is_media_type_in tests if one media type is in a list of others. Added static definitions for commonly used media types to res_pjsip.h. Changed several modules to use the new functions and static definitions. ASTERISK_29813 (not ready to close) Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9	5 years ago
George Joseph	1ce57621de	bundled_pjproject: Create generic pjsip_hdr_find functions ... pjsip_msg_find_hdr(), pjsip_msg_find_hdr_by_name(), and pjsip_msg_find_hdr_by_names() require a pjsip_msg to be passed in so if you need to search a header list that's not in a pjsip_msg, you have to do it yourself. This commit adds generic versions of those 3 functions that take in the actual header list head instead of a pjsip_msg so if you need to search a list of headers in something like a pjsip_multipart_part, you can do so easily. Change-Id: I6f2c127170eafda48e5e0d5d4d187bcd52b4df07	5 years ago
Sean Bright	101a72d048	say.c: Prevent erroneous failures with 'say' family of functions. ... A regression was introduced in ASTERISK~29531 that caused 'say' functions to fail with file lists that would previously have succeeded. This caused affected channels to hang up where previously they would have continued. We now explicitly check for the empty string to restore the previous behavior. ASTERISK-29859 #close Change-Id: Ia2e5769868e2792313c2d7c07996efe009c6f8d5	5 years ago
Naveen Albert	23c7c101ed	documentation: Document built-in system and channel vars ... Documentation for built-in special system and channel vars is currently outdated, and updating is a manual process since there is no XML documentation for these anywhere. This adds documentation for system vars to func_env and for channel vars to func_channel so that they appear along with the corresponding fields that would be accessed using a function. ASTERISK-29848 #close Change-Id: I6997f925c4a45fffe71321861f5898a8b7182fa9	5 years ago
Naveen Albert	84a596da5f	pbx_variables: add missing ASTSBINDIR variable ... Every config variable in the directories section of asterisk.conf currently has a counterpart built-in variable containing the value of the config option, except for the last one, astsbindir, which should have an ASTSBINDIR variable. However, the actual corresponding ASTSBINDIR variable is missing in pbx_variables.c. This adds the missing variable so that all the config options have their corresponding variable. ASTERISK-29847 #close Change-Id: I36006faf471825b36ebc8aa5e87a3bcb38d446fc	5 years ago
George Joseph	d6cdfb8204	bundled_pjproject: Make it easier to hack ... There are times when you need to troubleshoot issues with bundled pjproject or add new features that need to be pushed upstream but... * The source directory created by extracting the pjproject tarball is not scanned for code changes so you have to keep forcing rebuilds. * The source directory isn't a git repo so you can't easily create patches, do git bisects, etc. * Accidentally doing a make distclean will ruin your day by wiping out the source directory, and your changes. * etc. This commit makes that easier. See third-party/pjproject/README-hacking.md for the details. ASTERISK-29824 Change-Id: Idb1251040affdab31d27cd272dda68676da9b268	5 years ago