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Author	SHA1	Message	Date
George Joseph	bfc4d63d15	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	3 years ago
George Joseph	8d571ea6b5	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	3 years ago
Naveen Albert	386c5e495f	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	3 years ago
Naveen Albert	70f8ea0d1a	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	3 years ago
Naveen Albert	7ae8321925	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	3 years ago
Torrey Searle	7b15ced930	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	3 years ago
Kevin Harwell	851a759619	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	3 years ago
Sean Bright	ce91a0fdbc	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	3 years ago
Mike Bradeen	b79a571279	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	3 years ago
Mark Petersen	93d090147f	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	3 years ago
Luke Escude	5875c7bb6c	res_pjsip_sdp_rtp.c: Support keepalive for video streams. ... ASTERISK-28890 #close Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1	3 years ago
Michał Górny	23be22abf4	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	3 years ago
Michał Górny	2b490787eb	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	3 years ago
Michał Górny	dda02b8979	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	3 years ago
Michał Górny	6a879eea31	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	3 years ago
Michał Górny	710c8f8b29	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	3 years ago
Naveen Albert	27502b6dd2	frame.h: Fix spelling typo ... Fixes CNG description from "noice" to "noise". ASTERISK-29855 #close Change-Id: Ie7cbbd7d72b426693df7447384ff8700318cd36d	3 years ago
Naveen Albert	d35e292ae4	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	3 years ago
George Joseph	97ace6b816	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	3 years ago
George Joseph	b1dfc9c805	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	3 years ago
George Joseph	5d1407aa06	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	3 years ago
George Joseph	921ab52cf3	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	3 years ago
George Joseph	0d1b9e6baf	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	3 years ago
Sean Bright	65b2ddee26	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	3 years ago
Naveen Albert	5f59e0d36f	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	3 years ago
Naveen Albert	fbaf74bd3a	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	3 years ago
George Joseph	bc59b66de3	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	3 years ago
Sean Bright	0d62735f99	utils.c: Remove all usages of ast_gethostbyname() ... gethostbyname() and gethostbyname_r() are deprecated in favor of getaddrinfo() which we use in the ast_sockaddr family of functions. ASTERISK-29819 #close Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7	3 years ago
Naveen Albert	262a4053ff	say.conf: fix 12pm noon logic ... Fixes 12pm noon incorrectly returning 0/a.m. Also fixes a misspelling typo in the config. ASTERISK-29695 #close Change-Id: Ie40f9618636eb4c483b449bd707a5dcffca5c406	3 years ago
Sean Bright	3616dda066	pjproject: Fix incorrect unescaping of tokens during parsing ... ASTERISK-29664 #close Change-Id: I29dcde52e9faeaf2609c604eada61c6a9e49d8f5	3 years ago
Mark Petersen	dc7bcd68e4	app_queue.c: Support for Nordic syntax in announcements ... adding support for playing the correct en/et for nordic languages by adding 'n' for neuter gender in the relevant ast_say_number ASTERISK-29827 Change-Id: I03ebc827d2f0dc95132ab2f42799893c70edc5b1	3 years ago
Naveen Albert	138fbfa274	dsp: Add define macro for DTMF_MATRIX_SIZE ... Adds the macro DTMF_MATRIX_SIZE to replace the magic number 4 sprinkled throughout dsp.c. ASTERISK-29815 #close Change-Id: Ie3bddb92c6b16204ece0f758009e9490eb33b9ba	3 years ago
Naveen Albert	68f1e5d508	ami: Add AMI event for Wink ... Adds an AMI event for a wink frame. ASTERISK-29830 #close Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56	3 years ago
Naveen Albert	5b8d68d678	cli: Add module refresh command ... Adds a command to the CLI to unload and then load a module. This makes it easier to perform these operations which are often done subsequently to load a new version of a module. "module reload" already refers to reloading of configuration, so the name "refresh" is chosen instead. ASTERISK-29807 #close Change-Id: I595f6f11774a0de2565a1fba38da22309ce93a2c	3 years ago
Naveen Albert	80766059ef	app_mp3: Throw warning on nonexistent stream ... Currently, the MP3Player application doesn't emit a warning if attempting to play a stream which no longer exists. This can be a common scenario as many mp3 streams are valid at some point but can disappear at any time. Now a warning is thrown if attempting to play a nonexistent MP3 stream, instead of silently exiting. ASTERISK-29829 #close Change-Id: I53a0bf1ed1740166655eb66fe7675f6f808bf535	3 years ago
Naveen Albert	70bc0ff9d0	documentation: Add missing AMI documentation ... Adds missing documentation for some channel, bridge, and queue events. ASTERISK-24427 ASTERISK-29515 Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64	3 years ago
Kevin Harwell	1ddaedeaf5	tcptls.c: refactor client connection to be more robust ... The current TCP client connect code, blocks and does not handle EINTR error case. This patch makes the client socket non-blocking while connecting, ensures a connect does not immediately fail due to EINTR "errors", and adds a connect timeout option. The original client start call sets the new timeout option to "infinite", thus making sure old, orginal behavior is retained. ASTERISK-29746 #close Change-Id: I907571843a83e43c0742b95a64785f4411f02671	3 years ago
Naveen Albert	f7c4a3800c	app_sf: Add full tech-agnostic SF support ... Adds tech-agnostic support for SF signaling by adding SF sender and receiver applications as well as Dial integration. ASTERISK-29802 #close Change-Id: I7ec50752e9a661af639425e5d1e339f17411bcad	3 years ago
Steve Davies	a2ea233a6d	app_queue: Fix hint updates, allow dup. hints ... A previous patch for ASTERISK_29578 caused a 'leak' of extension state information across queues, causing the state of the first member of unrelated queues to be updated in addition to the correct member. Which queues and members depended on the order of queues in the iterator. Additionally, it is possible to use the same 'hint:' on multiple queue members, so the update cannot break out of the update loop early when a match is found. ASTERISK-29806 #close Change-Id: If2c1d1cc2a752afd9286d79710fc818596e7a7ad	3 years ago
Sean Bright	3fd12f1aa3	say.c: Honor requests for DTMF interruption. ... SayAlpha, SayAlphaCase, SayDigits, SayMoney, SayNumber, SayOrdinal, and SayPhonetic all claim to allow DTMF interruption if the SAY_DTMF_INTERRUPT channel variable is set to a truthy value, but we are failing to break out of a given 'say' application if DTMF actually occurs. ASTERISK-29816 #close Change-Id: I6a96e0130560831d2cb45164919862b9bcb6287e	3 years ago
Florentin Mayer	dd41572f99	res_pjsip_sdp_rtp: Preserve order of RTP codecs ... The ast_rtp_codecs_payloads functions do not preserve the order in which the payloads were specified on an incoming SDP media line. This leads to a problem with the codec negotiation functionality, as the format capabilities of the stream are extracted from the ast_rtp_codecs. This commit moves the ast_rtp_codec to ast_format conversion to the place where the order is still known. ASTERISK-28863 ASTERISK-29320 Change-Id: I3aabcfed3f379c36654f59c1872c313d0cb57e25	3 years ago
Joshua C. Colp	f9e67945da	bridge: Unlock channel during Local peer check. ... It's not safe to keep the channel locked while locking the peer Local channel, as it can result in a deadlock. This change unlocks it during this time but keeps the bridge locked to ensure nothing changes about the bridge. ASTERISK-29821 Change-Id: Ib68eb7037e5a479bcc2aceee77337cdde1fbdde6	3 years ago
Josh Soref	2b61440027	test_time.c: Tolerate DST transitions ... When test_timezone_watch runs very near a DST transition, two time zones that would otherwise be expected to report the same time can differ because of the DST transition. Instead of having the test fail when this happens, report the times, time zones, and dst flags. ASTERISK-29722 Change-Id: Id59bdac8b277e14343ccdf0c99b89e92f79f316a	3 years ago
George Joseph	7728210352	bundled_pjproject: Add more support for multipart bodies ... Adding upstream patch for pull request... https://github.com/pjsip/pjproject/pull/2920 --------------------------------------------------------------- sip_inv: Additional multipart support (#2919) sip_inv.c:inv_check_sdp_in_incoming_msg() deals with multipart message bodies in rdata correctly. In the case where early media is involved though, the existing sdp has to be retrieved from the last tdata sent in this transaction. This, however, always assumes that the sdp sent is in a non-multipart body. While there's a function to retrieve the sdp from multipart and non-multpart rdata bodies, no similar function for tdata exists. So... * The existing pjsip_rdata_get_sdp_info2 was refactored to find the sdp in any body, multipart or non-multipart, and from either an rdata or tdata. The new function is pjsip_get_sdp_info. This new function detects whether the pjsip_msg->body->data is the text representation of the sdp from an rdata or an existing pjmedia_sdp_session object from a tdata, or whether pjsip_msg->body is a multipart body containing either of the two sdp formats. * The exsting pjsip_rdata_get_sdp_info and pjsip_rdata_get_sdp_info2 functions are now wrappers that get the body and Content-Type header from the rdata and call pjsip_get_sdp_info. * Two new wrappers named pjsip_tdata_get_sdp_info and pjsip_tdata_get_sdp_info2 have been created that get the body from the tdata and call pjsip_get_sdp_info. * inv_offer_answer_test.c was updated to test multipart scenarios. ASTERISK-29804 Change-Id: I483c7c3d413280c9e247a96ad581278347f9c71b	4 years ago
Frederic Van Espen	cb44ceadec	ast_coredumper: Fix deleting results when output dir is set ... When OUTPUTDIR is set to another directory and the --delete-results-after is set, the resulting txt files are not deleted. ASTERISK-29794 #close Change-Id: I1c0071f6809a1e3f5cfc455d6eb08378bc0d7286	4 years ago
Naveen Albert	cfcbf0adad	pbx_variables: initialize uninitialized variable ... The variable cp4 in a variable substitution function can potentially be used without being initialized currently. This causes Asterisk to no longer compile. This initializes cp4 to NULL to make the compiler happy. ASTERISK-29803 #close Change-Id: I392579cbb76db2795d5820c9427cf55fbcee9e72	4 years ago
Mark Petersen	92cb1c0a59	app_queue.c: added DIALEDPEERNUMBER on outgoing channel ... added that we set DIALEDPEERNUMBER on the outgoing channels so it is avalible in b(content^extension^line) this add the same behaviour as Dial ASTERISK-29795 Change-Id: Icbc589ea2066f0c401a892bf478f6b2fd44e62f6	4 years ago
Kevin Harwell	1c389faa31	http.c: Add ability to create multiple HTTP servers ... Previously, it was only possible to have one HTTP server in Asterisk. With this patch it is now possible to have multiple HTTP servers listening on different addresses. Note, this behavior has only been made available through an API call from within the TEST_FRAMEWORK. Specifically, this feature has been added in order to allow unit test to create/start and stop servers, if one has not been enabled through configuration. Change-Id: Ic5fb5f11e62c019a1c51310f4667b32a4dae52f5	4 years ago
Naveen Albert	b951821eb7	app.c: Throw warnings for nonexistent options ... Currently, Asterisk doesn't throw warnings if options are passed into applications that don't accept them. This can confuse users if they're unaware that they are doing something wrong. This adds an additional check to parse_options so that a warning is thrown anytime an option is parsed that doesn't exist in the parsing application, so that users are notified of the invalid usage. ASTERISK-29801 #close Change-Id: Id029274a57135caca193c913307a63fd75e24679	4 years ago
Mark Petersen	4f06de7cf8	app_voicemail.c: Support for Danish syntax in VM ... added support for playing the correct plural sound file dependen on where you have 1 or multipe messages based on the existing SE/NO code ASTERISK-29797 Change-Id: I88aa814d02f3772bb80b474204b1ffb26fe438c2	4 years ago