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Author	SHA1	Message	Date
Michał Górny	d68d90c5be	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	4 years ago
Michał Górny	90d02cf0a3	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	4 years ago
Michał Górny	c8ef232d76	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	4 years ago
Michał Górny	7b1e5fa34a	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	4 years ago
Michał Górny	eef29d24e1	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	4 years ago
Naveen Albert	18c257b44d	frame.h: Fix spelling typo ... Fixes CNG description from "noice" to "noise". ASTERISK-29855 #close Change-Id: Ie7cbbd7d72b426693df7447384ff8700318cd36d	4 years ago
Naveen Albert	a9e9e15c3a	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	4 years ago
George Joseph	cc38ed9c21	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	4 years ago
George Joseph	f55886a72c	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	4 years ago
George Joseph	59cf9f0047	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	4 years ago
George Joseph	b59bd3d3e4	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	4 years ago
George Joseph	f10947ecc2	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	4 years ago
Sean Bright	3fd47840c9	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	4 years ago
Naveen Albert	e006d2d2a6	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	4 years ago
Naveen Albert	707f32170c	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	4 years ago
George Joseph	3f093b8dda	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	4 years ago
Sean Bright	0bbef4d8c5	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	4 years ago
Naveen Albert	54f2f1e027	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	4 years ago
Sean Bright	ee69441fbd	pjproject: Fix incorrect unescaping of tokens during parsing ... ASTERISK-29664 #close Change-Id: I29dcde52e9faeaf2609c604eada61c6a9e49d8f5	4 years ago
Mark Petersen	dea71ddbbf	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	4 years ago
Naveen Albert	6d7161820e	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	4 years ago
Naveen Albert	f133ae6ca2	ami: Add AMI event for Wink ... Adds an AMI event for a wink frame. ASTERISK-29830 #close Change-Id: I83e426de5e37baed79a4dbcc91e9e8d030ef1b56	4 years ago
Naveen Albert	1c2f311ba3	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	4 years ago
Naveen Albert	775c371d09	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	4 years ago
Naveen Albert	b37feb42ae	documentation: Add missing AMI documentation ... Adds missing documentation for some channel, bridge, and queue events. ASTERISK-24427 ASTERISK-29515 Change-Id: I92b06b88c8cadc0155f95ebe3e870b3e795a8c64	4 years ago
Kevin Harwell	06f9227ac5	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	4 years ago
Naveen Albert	dd6df42534	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	4 years ago
Steve Davies	16a63027c0	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	4 years ago
Sean Bright	4fe94bab09	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	4 years ago
Florentin Mayer	4e204db2bf	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	4 years ago
Joshua C. Colp	d83a46869e	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	4 years ago
Josh Soref	a5cdee36a7	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	4 years ago
George Joseph	0cf4e325aa	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	965f4abd9a	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	bb27d5e1fe	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	3d71bcd2f4	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	05afa061f5	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	030f7d4131	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	a4c42e70c1	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
Naveen Albert	86bc3eef99	strings: Fix enum names in comment examples ... The enum values for ast_strsep_flags includes AST_STRSEP_STRIP. However, some comments reference AST_SEP_STRIP, which doesn't exist. This fixes these comments to use the correct value. ASTERISK-29800 #close Change-Id: If7bbd0c0e6226a211d25ddf9d1629347e2674943	4 years ago
Naveen Albert	c6410dc4ed	configs: Updates to sample configs ... Includes some minor updates to extensions.conf and iax.conf. In particular, the demonstration of macros in extensions.conf is removed, as Macro is deprecated and will be removed soon. These examples have been replaced with examples demonstrating the usage of Gosub instead. The older exten => ...,n syntax is also mostly replaced with the same keyword to demonstrate the newer, more concise way of defining extensions. IAXTEL no longer exists, so this example is replaced with something more generic. Some documentation is also added to extensions.conf and iax.conf to clarify some of the new expanded encryption capabilities with IAX2. ASTERISK-29758 #close Change-Id: I04fba9671aa1ee9ba1bd5027061f80bbe38e7b46	4 years ago
Naveen Albert	2f1eb56116	app_sendtext: Add ReceiveText application ... Adds a ReceiveText application that can be used in conjunction with SendText. Currently, there is no way in Asterisk to receive text in the dialplan (or anywhere else, really). This allows for Asterisk to be the recipient of text instead of just the sender. ASTERISK-29759 #close Change-Id: Ica2c354a42bff69f323a0493d3a7cd0fb129d52d	4 years ago
Naveen Albert	c6309af560	pbx_variables: Increase parsing capabilities of MSet ... Currently MSet can only parse a maximum of 24 variables. If more variables are provided to MSet, the 24th variable will simply contain the remainder of the string and the remaining variables thereafter will never get set. This increases the number of variables that can be parsed in one go from 24 to 99. Additionally, documentation is added since this limitation is currently undocumented and is confusing to users who encounter this limitation. ASTERISK-29766 #close Change-Id: I3fe35b462dedec0a452fd9ea7f92c920a3939f16	4 years ago
Naveen Albert	3108457d8f	chan_sip: Fix crash when accessing RURI before initiating outgoing call ... Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before initiating an outgoing call will cause Asterisk to crash. This is because a null field is accessed, resulting in an offset from null and subsequent memory access violation. Since RURI is not guaranteed to exist, we now check if the base pointer is non-null before calculating an offset. ASTERISK-29772 Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83	4 years ago
Naveen Albert	c0cdaf0246	func_json: Adds JSON_DECODE function ... Adds the JSON_DECODE function for parsing JSON in the dialplan. JSON parsing already exists in the Asterisk core and is used for many different things. This function exposes the basic parsing capability to the user in the dialplan, for instance, in conjunction with CURL for using API responses. ASTERISK-29706 #close Change-Id: Iea60c49a7358dfdc2db60803cdc9a742f808ba2c	4 years ago
Sean Bright	087f25d3fd	CHANGES: Correct reference to configuration file. ... Change-Id: I22a788ebf11168fff7fbf9ea956ebcd705ab63dd	4 years ago
Naveen Albert	cc1418ef47	pbx: Add variable substitution API for extensions ... Currently, variable substitution involving dialplan extensions is quite clunky since it entails obtaining the current dialplan location, backing it up, storing the desired variables for substitution on the channel, performing substitution, then restoring the original location. In addition to being clunky, things could also go wrong if an async goto were to occur and change the dialplan location during a substitution. Fundamentally, there's no reason it needs to be done this way, so new API is added to allow for directly passing in the dialplan location for the purposes of variable substitution so we don't need to mess with the channel information anymore. Existing API is not changed. ASTERISK-29745 #close Change-Id: I23273bf27fa0efb64a606eebf9aa8e2f41a065e4	4 years ago
Naveen Albert	2b2b708d43	app_mf: Add full tech-agnostic MF support ... Adds tech-agnostic support for MF signaling by adding MF sender and receiver applications as well as Dial integration. ASTERISK-29496-mf #do-not-close Change-Id: I61962b359b8ec4cfd05df877ddf9f5b8f71927a4	4 years ago
Alexander Traud	55dd77b921	xmldoc: Avoid whitespace around value for parameter/required. ... Otherwise, the value 'false' was not found in the enumerated set of the XML DTD for the XML attribute 'required' in the XML element 'parameter'. Therefore, DTD validation of the runtime XML failed. ASTERISK-29790 Change-Id: Id13f230ad65a70dd8c2e3ae9ac85d1e841aed03e	4 years ago
Alexander Traud	4b6c72572c	progdocs: Fix Doxygen left-overs. ... Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016	4 years ago