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Author	SHA1	Message	Date
Philip Prindeville	70a958a051	logger: workaround woefully small BUFSIZ in MUSL ... MUSL defines BUFSIZ as 1024 which is not reasonable for log messages. More broadly, BUFSIZ is the amount of buffering stdio.h does, which is arbitrary and largely orthogonal to what logging should accept as the maximum message size. ASTERISK-29928 Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com> Change-Id: Iaa49fbbab029c64ae3d95e4b18270e0442cce170	4 years ago
Naveen Albert	297ceeb9e4	app_dial: Document DIALSTATUS return values. ... Adds documentation for all of the possible return values for the DIALSTATUS variable in the Dial application. ASTERISK-25716 Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0	4 years ago
Sean Bright	b15d2f9e4b	stasis_recording: Perform a complete match on requested filename. ... Using the length of a file found on the filesystem rather than the file being requested could result in filenames whose names are substrings of another to be erroneously matched. We now ensure a complete comparison before returning a positive result. ASTERISK-29960 #close Change-Id: Id3ffc77681b9b75b8569062f3d952a128a21c71a	4 years ago
Sean Bright	5a30f0c6fe	download_externals: Use HTTPS for downloads ... ASTERISK-29980 #close Change-Id: I7b347665822ea2774dd322276c09be67914d2065	4 years ago
Alexei Gradinari	4027626709	res_pjsip_pubsub: RLS 'uri' list attribute mismatch with SUBSCRIBE request ... When asterisk generates the RLMI part of NOTIFY request, the asterisk uses the local contact uri instead of the URI to which the SUBSCRIBE request is sent. Because of this mismatch some IP phones (for example Cisco 5XX) ignore this list. According https://datatracker.ietf.org/doc/html/rfc4662#section-5.2 The first mandatory <list> attribute is "uri", which contains the uri that corresponds to the list. Typically, this is the URI to which the SUBSCRIBE request was sent. https://datatracker.ietf.org/doc/html/rfc4662#section-5.3 The "uri" attribute identifies the resource to which the <resource> element corresponds. Typically, this will be a SIP URI that, if subscribed to, would return the state of the resource. This patch makes asterisk to generate URI using SUBSCRIBE request URI. ASTERISK-29961 #close Change-Id: I1fcfc08fd589677f40608c59a4e143c45ee05f6c	4 years ago
Naveen Albert	c5b49f1405	pbx_builtins: Add missing options documentation ... BackGround and WaitExten both accept options that are not currently documented. This adds documentation for these options to the xml documentation for each application. ASTERISK-29967 #close Change-Id: If812a9f1ccbba3e4d427a0e7a6dea923c2f905f7	4 years ago
Asterisk Development Team	96f98908f9	Update CHANGES and UPGRADE.txt for 19.3.0	4 years ago
Ben Ford	0c07f94c00	AMI: Bump version for 19.3.0. ... Change-Id: Ia41acd3f4f9589e8d3ecda3193764c83dade6186	4 years ago
Alexei Gradinari	49928fee53	res_pjsip_pubsub: update RLS to reflect the changes to the lists ... This patch makes the Resource List Subscriptions (RLS) dynamic. The asterisk updates the current subscriptions to reflect the changes to the list on the subscriptions refresh. If list items are added, removed, updated or do not exist anymore, the asterisk regenerates the resource list. ASTERISK-29906 #close Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3	4 years ago
Boris P. Korzun	00a7fa985e	res_config_pgsql: Add text-type column check in require_pgsql() ... Omit "unsupported column type 'text'" warning in logs while using text-type column in the PgSQL backend. ASTERISK-29924 #close Change-Id: I48061a7d469426859670db07f1ed8af1eb814712	4 years ago
Kfir Itzhak	5edbc54c54	app_queue: Add QueueWithdrawCaller AMI action ... This adds a new AMI action called QueueWithdrawCaller. This AMI action makes it possible to withdraw a caller from a queue, in a safe and a generic manner. This can be useful for retrieving a specific call and dispatching it to a specific extension. It works by signaling the caller to exit the queue application whenever it can. Therefore, it is not guaranteed that the call will leave the queue. ASTERISK-29909 #close Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec	4 years ago
Kevin Harwell	b81db677c9	AST-2022-006: pjproject - unconstrained malformed multipart SIP message ... ASTERISK-29945 #close Change-Id: Ic58957afc453195d53c2bd25c905df3d91d1abe6	4 years ago
Kevin Harwell	f9cd83ff35	AST-2022-005: pjproject - undefined behavior after freeing a dialog set ... ASTERISK-29945 #close Change-Id: Ia8ce6d82b115c82c1138747c72a0adcaa42b718c	4 years ago
Kevin Harwell	ac8de6e31d	AST-2022-004: pjproject - possible integer underflow on STUN message ... ASTERISK-29945 #close Change-Id: I721cd254e4f8aa6d3a97a37529cca53519694c54	4 years ago
George Joseph	1dc792a312	xml.c, config,c: Add stylesheets and variable list string parsing ... Added functions to open, close, and apply XML Stylesheets to XML documents. Although the presence of libxslt was already being checked by configure, it was only happening if xmldoc was enabled. Now it's checked regardless. Added ability to parse a string consisting of comma separated name/value pairs into an ast_variable list. The reverse of ast_variable_list_join(). Change-Id: I1e1d149be22165a1fb8e88e2903a36bba1a6cf2e	4 years ago
George Joseph	d00108df72	xmldoc: Fix issue with xmlstarlet validation ... Added the missing xml-stylesheet and Xinclude namespace declarations in pjsip_config.xml and pjsip_manager.xml. Updated make_xml_documentation to show detailed errors when xmlstarlet is the validator. It's now run once with the '-q' option to suppress harmless/expected messages and if it actually fails, it's run again without '-q' but with '-e' to show the actual errors. Change-Id: I4bdc9d2ea6741e8d2e5eb82df60c68ccc59e1f5e	4 years ago
George Joseph	38df1c35ee	core: Config and XML tweaks needed for geolocation ... Added: Replace a variable in a list: int ast_variable_list_replace_variable(struct ast_variable **head, struct ast_variable *old, struct ast_variable *new); Added test as well. Create a "name=value" string from a variable list: 'name1="val1",name2="val2"', etc. struct ast_str *ast_variable_list_join( const struct ast_variable *head, const char *item_separator, const char *name_value_separator, const char *quote_char, struct ast_str **str); Added test as well. Allow the name of an XML element to be changed. void ast_xml_set_name(struct ast_xml_node *node, const char *name); Change-Id: I330a5f63dc0c218e0d8dfc0745948d2812141ccb	4 years ago
George Joseph	e29cd99975	Makefile: Allow XML documentation to exist outside source files ... Moved the xmldoc build logic from the top-level Makefile into its own script "make_xml_documentation" in the build_tools directory. Created a new utility script "get_sourceable_makeopts", also in the build_tools directory, that dumps the top-level "makeopts" file in a format that can be "sourced" from shell sscripts. This allows scripts to easily get the values of common make build variables such as the location of the GREP, SED, AWK, etc. utilities as well as the AST* and library *_LIB and *_INCLUDE variables. Besides moving logic out of the Makefile, some optimizations were done like removing "third-party" from the list of subdirectories to be searched for documentation and changing some assignments from "=" to ":=" so they're only evaluated once. The speed increase is noticeable. The makeopts.in file was updated to include the paths to REALPATH and DIRNAME. The ./conifgure script was setting them but makeopts.in wasn't including them. So... With this change, you can now place documentation in any"c" source file AND you can now place it in a separate XML file altogether. The following are examples of valid locations: res/res_pjsip.c Using the existing /*** DOCUMENTATION ***/ fragment. res/res_pjsip/pjsip_configuration.c Using the existing /*** DOCUMENTATION ***/ fragment. res/res_pjsip/pjsip_doc.xml A fully-formed XML file. The "configInfo", "manager", "managerEvent", etc. elements that would be in the "c" file DOCUMENTATION fragment should be wrapped in proper XML. Example for "somemodule.xml": <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE docs SYSTEM "appdocsxml.dtd"> <docs> <configInfo> ... </configInfo> </docs> It's the "appdocsxml.dtd" that tells make_xml_documentation that this is a documentation XML file and not some other XML file. It also allows many XML-capable editors to do formatting and validation. Other than the ".xml" suffix, the name of the file is not significant. As a start... This change also moves the documentation that was in res_pjsip.c to 2 new XML files in res/res_pjsip: pjsip_config.xml and pjsip_manager.xml. This cut the number of lines in res_pjsip.c in half. :) Change-Id: I486c16c0b5a44d7a8870008e10c941fb19b71ade	4 years ago
George Joseph	ca18e68470	build: Refactor the earlier "basebranch" commit ... Recap from earlier commit: 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" or a work branch based on that "development" branch. 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. It also constructs the URLs for downloading external modules with that version, which will fail. Fast-forward: The earlier attempt at adding a "basebranch" variable to .gitreview didn't work out too well in practice because changes were made to .gitreview, which is a checked-in file. So, if you wanted to rebase your work branch on the base branch, rebase would attempt to overwrite your .gitreview with the one from the base branch and complain about a conflict. This is a slighltly different approach that adds three methods to determine the mainline branch: 1. --- MAINLINE_BRANCH from the environment If MAINLINE_BRANCH is already set in the environment, that will be used. This is primarily for the Jenkins jobs. 2. --- .develvars Instead of storing the basebranch in .gitreview, it can now be stored in a non-checked-in ".develvars" file and keyed by the current branch. So, if you were working on a branch named "new-feature-work" based on "development/16/new-feature" and wanted to push to that branch in Gerrit but wanted to pull the external modules for 16, you'd create the following .develvars file: [branch "new-feature-work"] mainline-branch = 16 The .gitreview file would still look like: [gerrit] defaultbranch=development/16/new-feature ...which would cause any reviews pushed from "new-feature-work" to go to the "development/16/new-feature" branch in Gerrit. The key is that the .develvars file is NEVER checked in (it's been added to .gitignore). 3. --- Well Known Development Branch If you're actually working in a branch named like "development/<mainline_branch>/some-feature", the mainline branch will be parsed from it. 4. --- .gitreview If none of the earlier conditions exist, the .gitreview "defaultbranch" variable will be used just as before. Change-Id: I1cdeeaa0944bba3f2e01d7a2039559d0c266f8c9	4 years ago
Joshua C. Colp	3d30f68fbc	jansson: Update bundled to 2.14 version. ... ASTERISK-29353 Change-Id: I4ea43eda1691565563a4c03ef37166952d211b2b	4 years ago
Naveen Albert	8a94702386	ami: Allow events to be globally disabled. ... The disabledevents setting has been added to the general section in manager.conf, which allows users to specify events that should be globally disabled and not sent to any AMI listeners. This allows for processing of these AMI events to end sooner and, for frequent AMI events such as Newexten which users may not have any need for, allows them to not be processed. Additionally, it also cleans up core debug as previously when debug was 3 or higher, the debug was constantly spammed by "Analyzing AMI event" messages along with a complete dump of the event contents (often for Newexten). ASTERISK-29853 #close Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205	4 years ago
Naveen Albert	ff603d16c7	func_channel: Add lastcontext and lastexten. ... Adds the lastcontext and lastexten channel fields to allow users to access previous dialplan execution locations. ASTERISK-29840 #close Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307	4 years ago
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	5 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	5 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	5 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	5 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	5 years ago
Sean Bright	dfbb547f6a	res_pjsip.c: Correct minor typos in 'realm' documentation. ... Change-Id: I886936b808def5540d40071321e72f6bfa19063a	5 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	5 years ago
Asterisk Development Team	3d390c4df7	Update CHANGES and UPGRADE.txt for 19.2.0	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 years ago