asterisk

Commit Graph

Author	SHA1	Message	Date
Naveen Albert	67a87c01ff	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	9e975d4c18	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	5fdf6e4e18	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	2f2af26e96	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
Sean Bright	96c2a7932d	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
George Joseph	6dbed99435	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
Naveen Albert	902c70530c	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
Naveen Albert	1859388341	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
George Joseph	d014cedf5d	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	6060b9ccb0	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	75b9b93ca3	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	85881d84dc	pjproject: Fix incorrect unescaping of tokens during parsing ASTERISK-29664 #close Change-Id: I29dcde52e9faeaf2609c604eada61c6a9e49d8f5	4 years ago
Mark Petersen	ac9607d03b	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	57d5964d77	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	06eff7d5d0	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	ef00ff4e1e	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	dc5d70f29b	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	6eeb7ab7d0	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	a3fcda5a05	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	2ed004846a	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	076d502694	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	3f398c319f	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
Joshua C. Colp	a363153fa8	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	d8d4ac87a5	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	f2cfce147b	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	144e4772e4	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	ffb98d764c	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	0dbf313a76	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	b37fd02201	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	1e7bcc27df	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	ff0b048543	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	3c08ce4f72	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	216a18b291	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	85dc5214e3	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	3629b7f825	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	1ac89e0875	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	0eaa214be9	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	33f0500359	CHANGES: Correct reference to configuration file. Change-Id: I22a788ebf11168fff7fbf9ea956ebcd705ab63dd	4 years ago
Naveen Albert	bf28c2901d	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	9eba9f44d6	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	a7688c0216	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	ca64087486	progdocs: Fix Doxygen left-overs. Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016	4 years ago
Alexander Traud	090c56d80c	xmldoc: Correct definition for XML element 'matchInfo'. ASTERISK-29791 Change-Id: I7c656498427fcadd0a5d61a54ff67e6036609725	4 years ago
Alexander Traud	b49ae92fe7	progdocs: Update Makefile. In developer mode, use internal documentation as well. This should produce no warnings. Fix yours! In noisy mode, output all possible warnings of Doxygen. This creates zillion of warnings. Double-check your current module! Any warnings are in the file './doxygen.log'. Beside that, this change avoids deprecated parameters because the configuration file for Doxygen contains only those parameters which differ from the default. This avoids the need to update the file on each run. Furthermore, it adds AST_VECTOR to be expanded. Finally, the default name for that file is Doxyfile. Therefore, let us use that! ASTERISK-26991 ASTERISK-20259 Change-Id: I4129092a199d5e24c319a09cd088614b121015af	4 years ago
Alexander Traud	65387cd442	res_pjsip_sdp_rtp: Do not warn on unknown sRTP crypto suites. res_sdp_crypto_parse_offer(.) emits many log messages already. ASTERISK-29785 Change-Id: I1a191ebe4fec1102946d4e31887e5197ca02dfe8	4 years ago
Sean Bright	00c3d60881	channel: Short-circuit ast_channel_get_by_name() on empty arg. We know that passing a NULL or empty argument to ast_channel_get_by_name() will never result in a matching channel and will always result in an error being emitted, so just short-circuit out in that case. ASTERISK-28219 #close Change-Id: I88eadc748e9c6996fc17467b0a05881bbfd00bce	4 years ago
Mike Bradeen	a64129943b	res_rtp_asterisk: Addressing possible rtp range issues res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined that rtpstart was configured to be an odd value. Also adding a loop counter to prevent a possible infinite loop when looking for a free port. ASTERISK-27406 Change-Id: I90f07deef0716da4a30206e9f849458b2dbe346b	4 years ago
Mark Petersen	caa12fecb1	apps/app_dial.c: HANGUPCAUSE reason code for CANCEL is set to AST_CAUSE_NORMAL_CLEARING changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING ASTERISK-28053 Reported by: roadkill Change-Id: Ib653aec2282f55b59d87484391cc07c8e6612b89	4 years ago
Alexander Traud	14736ae6d4	res: Fix for Doxygen. These are the remaining issues found in /res. ASTERISK-29761 Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d	4 years ago
Dustin Marquess	3162f6dfdd	res_fax_spandsp: Add spandsp 3.0.0+ compatibility Newer versions of spandsp did refactoring of code to add new features like color FAXing. This refactoring broke backwards compatibility. Add support for the new version while retaining support for 0.0.6. ASTERISK-29729 #close Change-Id: I3bd74550604ebcf0304528d647fa39abc62fbaa1	4 years ago

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