asterisk

Commit Graph

Author	SHA1	Message	Date
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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 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	5 years ago
Naveen Albert	54761a41cd	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	5 years ago
Naveen Albert	8ec13f06de	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	5 years ago
Naveen Albert	5c67a991c2	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	5 years ago
Naveen Albert	97f400100c	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	5 years ago
Naveen Albert	b64e894650	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	5 years ago
Naveen Albert	c3ff464864	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	5 years ago
Naveen Albert	23a4a12420	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	5 years ago
Sean Bright	6a6967bf0c	CHANGES: Correct reference to configuration file. Change-Id: I22a788ebf11168fff7fbf9ea956ebcd705ab63dd	5 years ago
Naveen Albert	ee9eef492c	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	5 years ago
Alexander Traud	67c4661fb0	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	5 years ago
Alexander Traud	826233b550	progdocs: Fix Doxygen left-overs. Change-Id: I5b5cf9c9cbbe00ba8b379a8d162ac67445d39016	5 years ago
Alexander Traud	12c45dd6a2	xmldoc: Correct definition for XML element 'matchInfo'. ASTERISK-29791 Change-Id: I7c656498427fcadd0a5d61a54ff67e6036609725	5 years ago
Alexander Traud	f3b29c6aa8	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	5 years ago
Alexander Traud	f6df28ce87	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	5 years ago
Sean Bright	e9cac5f4bf	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	5 years ago
Mike Bradeen	59fcd1e7e2	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	5 years ago
Mark Petersen	a8b2692836	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	5 years ago
Alexander Traud	a85f2bf34d	res: Fix for Doxygen. These are the remaining issues found in /res. ASTERISK-29761 Change-Id: I572e6019c422780dde5ce8448b6c85c77af6046d	5 years ago
Dustin Marquess	e93fb874b4	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	5 years ago
Alexander Traud	9440f6ec58	main: Fix for Doxygen. ASTERISK-29763 Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e	5 years ago
Alexander Traud	cc025026b7	progdocs: Fix for Doxygen, the hidden parts. ASTERISK-29779 Change-Id: If338163488498f65fa7248b60e80299c0a928e4b	5 years ago
Alexander Traud	affe7ee879	progdocs: Fix grouping for latest Doxygen. Since Doxygen 1.8.16, a special comment block is required. Otherwise (pure C comment), the group command is ignored. Additionally, several unbalanced group commands were fixed. ASTERISK-29732 Change-Id: I4687857b9d56e6f44fd440b73af156691660202e	5 years ago
Naveen Albert	24a04054ad	documentation: Standardize examples Most examples in the XML documentation use the example tag to demonstrate examples, which gets parsed specially in the Wiki to make it easier to follow for users. This fixes a few modules to use the example tag instead of vanilla para tags to bring them in line with the standard syntax. ASTERISK-29777 #close Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b	5 years ago
Sean Bright	2478bfcff9	config.c: Prevent UB in ast_realtime_require_field. A backend's implementation of the realtime 'require' function may call va_arg() and then fail, leaving the va_list in an undefined state. Pass a copy of the va_list instead. ASTERISK-29771 #close Change-Id: I555565a72af84e96d49f62fe8cb66ba5a78461f4	5 years ago
Naveen Albert	d374d63ef8	app_voicemail: Refactor email generation functions Refactors generic functions used for email generation into utils.c so that they can be used by multiple modules, including app_voicemail and app_minivm, to avoid code duplication. ASTERISK-29715 #close Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee	5 years ago
Alexander Traud	ecffdab059	stir/shaken: Avoid a compiler extension of GCC. ASTERISK-29776 Change-Id: I86e5eca66fb775a5744af0c929fb269e70575a73	5 years ago
Alexander Traud	1230369b71	progdocs: Remove outdated references in doxyref.h. ASTERISK-29773 Change-Id: Ica93160d9158cc0e80c5fda829b80d1b49a6b9b9	5 years ago
Jaco Kroon	4b3c75ca31	logger: use __FUNCTION__ instead of __PRETTY_FUNCTION__ This avoids a few long-name overflows, at the cost of less instructive names in the case of C++ (specifically overloaded functions and class methods). This in turn is offset against the fact that we're logging the filename and line numbers in any case. Change-Id: I54101a0bb5f8cb9ef63ec12c5e0d4c8edafff9ed Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Alexander Traud	38f9000fcb	xmldoc: Fix for Doxygen. ASTERISK-29765 Change-Id: I654ba0debe8351038d4433716434a09370f04c9d	5 years ago
Mike Bradeen	4a4f1a5c9a	astobj2.c: Fix core when ref_log enabled In the AO2_ALLOC_OPT_LOCK_NOLOCK case the referenced obj structure is freed, but is then referenced later if ref_log is enabled. The change is to store the obj->priv_data.options value locally and reference it instead of the value from the freed obj ASTERISK-29730 Change-Id: I60cc5dc1f5a4330e7ad56976fc38a42de0ab6072	5 years ago
Alexander Traud	726d6dd166	channels: Fix for Doxygen. ASTERISK-29762 Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7	5 years ago
Joshua C. Colp	3a4c9ec0e2	bridge: Deny full Local channel pair in bridge. Local channels are made up of two pairs - the 1 and 2 sides. When a frame goes in one side, it comes out the other. Back and forth. When both halves are in a bridge this creates an infinite loop of frames. This change makes it so that bridging no longer allows both of these sides to exist in the same bridge. ASTERISK-29748 Change-Id: I29928b6de87cd9be996a77daccefd7c360fef651	5 years ago
Naveen Albert	4468fc11d6	res_tonedetect: Add call progress tone detection Makes basic call progress tone detection available in a tech-agnostic manner with the addition of the ToneScan application. This can determine if the channel has encountered a busy signal, SIT tones, dial tone, modem, fax machine, etc. A few basic async progress tone detect options are also added to the TONE_DETECT function. ASTERISK-29720 #close Change-Id: Ia02437e0450473031e294798b8cb421fb8f24e90	5 years ago
Boris P. Korzun	f6aed7b8d1	rtp_engine: Add type field for JSON RTCP Report stasis messages ASTERISK-29727 #close Change-Id: I2eca8aeb591cb63ac2238d08eab662367453cb82	5 years ago
Alexander Traud	00fc7212bd	odbc: Fix for Doxygen. ASTERISK-29754 Change-Id: Ia09eb68d283d201d9a6fbeccfc0efe83fe0502a5	5 years ago
Alexander Traud	241dbb1ec0	parking: Fix for Doxygen. ASTERISK-29753 Change-Id: I7a61974584f6169502e6860fc711919fe7bbfaa7	5 years ago

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33180 Commits (1ddaedeaf5b4307643d347ff8fbf485fbf8e2834) All Branches Search

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