asterisk

Commit Graph

Author	SHA1	Message	Date
Yury Kirsanov	e9355e66d1	bridge_simple.c: Unhold channels on join simple bridge. Patch provided inline by Yury Kirsanov on the linked issue and approved by Josh Colp. ASTERISK-29253 #close Change-Id: I5b9ccc67ebf06e875ed061d9e7fc21f47b0a4e1f	3 years ago
Kevin Harwell	272bac70dd	res_aeap & res_speech_aeap: Add Asterisk External Application Protocol Add framework to connect to, and read and write protocol based messages from and to an external application using an Asterisk External Application Protocol (AEAP). This has been divided into several abstractions: 1. transport - base communication layer (currently websocket only) 2. message - AEAP description and data (currently JSON only) 3. transaction - links/binds requests and responses 4. aeap - transport, message, and transaction handler/manager This patch also adds an AEAP implementation for speech to text. Existing speech API callbacks for speech to text have been completed making it possible for Asterisk to connect to a configured external translator service and provide audio for STT. Results can also be received from the external translator, and made available as speech results in Asterisk. Unit tests have also been created that test the AEAP framework, and also the speech to text implementation. ASTERISK-29726 #close Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2	3 years ago
Maximilian Fridrich	53a3af6321	app_dial: Flip stream direction of outgoing channel. When executing dial, the topology of the incoming channel is cloned and used for the outgoing channel. This creates issues when an incoming stream is sendonly or recvonly as the stream state of the outgoing channel will be the same as the stream state of the incoming channel. Now the stream state is flipped for the outgoing stream in dial_exec_full if the incoming stream topology is recvonly or sendonly. ASTERISK-29655 Reported by: Michael Auracher ASTERISK-29638 Reported by: Michael Auracher Change-Id: I294dc834ac9a5f048b101b691669959e9df630e1	3 years ago
Ben Ford	f593b1e93b	res_pjsip_stir_shaken.c: Fix enabled when not configured. There was an issue with the conditional where STIR/SHAKEN would be enabled even when not configured. It has been changed to ensure that if a profile does not exist and stir_shaken is not set in pjsip.conf, then the conditional will return from the function without performing STIR/SHAKEN operations. ASTERISK-30024 Change-Id: I41286a3d35b033ccbfbe4129427a62cb793a86e6	3 years ago
Joshua C. Colp	fdc1c750f3	res_pjsip: Always set async_operations to 1. The async_operations setting on a transport configures how many simultaneous incoming packets the transport can handle when multiple threads are polling and waiting on the transport. As we only use a single thread this was needlessly creating incoming packets when set to a non-default value, wasting memory. ASTERISK-30006 Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f	3 years ago
Sean Bright	b1e0527bbd	config.h: Don't use C++ keywords as argument names. ASTERISK-30021 #close Change-Id: I70eb59b782a4946b979942e21422746b7563029c	3 years ago
Joshua C. Colp	283b09cf70	cdr_adaptive_odbc: Add support for SQL_DATETIME field type. ASTERISK-30023 Change-Id: I0e1697f6af044e9eab7e07bbaeeffd1bb68ac34a	3 years ago
Joshua C. Colp	b3f39be0cc	pjsip: Increase maximum number of format attributes. Chrome has added more attributes, causing the limit to be exceeded. This raises it up some more. ASTERISK-30015 Change-Id: I964957c005c4e6f7871b15ea1ccd9b4659c7ef32	3 years ago
Ben Ford	0724b767a3	AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header. Adds a new configuration option, stir_shaken_profile, in pjsip.conf that can be specified on a per endpoint basis. This option will reference a stir_shaken_profile that can be configured in stir_shaken.conf. The type of this option must be 'profile'. The stir_shaken option can be specified on this object with the same values as before (attest, verify, on), but it cannot be off since having the profile itself implies wanting STIR/SHAKEN support. You can also specify an ACL from acl.conf (along with permit and deny lines in the object itself) that will be used to limit what interfaces Asterisk will attempt to retrieve information from when reading the Identity header. ASTERISK-29476 Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406	3 years ago
Ben Ford	8f3dd86b8d	AST-2022-001 - res_stir_shaken/curl: Limit file size and check start. Put checks in place to limit how much we will actually download, as well as a check for the data we receive at the start to ensure it begins with what we would expect a certificate to begin with. ASTERISK-29872 Change-Id: Ifd3c6b8bd52b8b6192a04166ccce4fc8a8000b46	3 years ago
Joshua C. Colp	4aedaaadeb	func_odbc: Add SQL_ESC_BACKSLASHES dialplan function. Some databases depending on their configuration using backslashes for escaping. When combined with the use of ' this can result in a broken func_odbc query. This change adds a SQL_ESC_BACKSLASHES dialplan function which can be used to escape the backslashes. This is done as a dialplan function instead of being always done as some databases do not require this, and always doing it would result in incorrect data being put into the database. ASTERISK-29838 Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d	3 years ago
Naveen Albert	b87c5f5124	app_mf, app_sf: Return -1 if channel hangs up. The ReceiveMF and ReceiveSF applications currently always return 0, even if a channel has hung up. The call will still end but generally applications are expected to return -1 if the channel has hung up. We now return -1 if a hangup occured to bring this behavior in line with this norm. This has no functional impact, but merely increases conformity with how these modules interact with the PBX core. ASTERISK-29951 #close Change-Id: I234d755050ab8ed58f197c6925b968ba26b14033	3 years ago
Naveen Albert	ede4e2099f	app_queue: Add music on hold option to Queue. Adds the m option to the Queue application, which allows a music on hold class to be specified at runtime which will override the class configured in queues.conf. This option functions like the m option to Dial. ASTERISK-29876 #close Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7	3 years ago
Naveen Albert	da44b848f5	app_meetme: Emit warning if conference not found. Currently, if a user tries to access a non-dynamic MeetMe conference and the conference is not found, the call simply silent hangs up. There is no indication to the user that anything went wrong at all. This changes the relevant debug message to a warning so that the user is notified of this invalidity. ASTERISK-29954 #close Change-Id: Iebcfae3755d00f2150d676ee211c57bc59530048	3 years ago
Naveen Albert	94df607771	build: Remove obsolete leftover build references. Removes some leftover build and config references to modules that have since been removed from Asterisk. ASTERISK-29935 #close Change-Id: Iaefc73a23f4b2de3c6c14d928050135b6d0ef6af	3 years ago
Kevin Harwell	0e31df6c93	res_pjsip_header_funcs: wrong pool used tdata headers When adding headers to an outgoing request the headers were cloned using the dialog's pool when they should have been cloned using tdata's pool. Under certain circumstances it was possible for the dialog object, and its pool to be freed while tdata is still active and available. Thus the cloned header "disappeared", and when tdata tried to later access it a crash would occur. This patch makes it so all added headers are cloned appropriately using tdata's pool. ASTERISK-29411 #close ASTERISK-29535 #close Change-Id: I9852025b5ee93ce1c038209150ee9dba1e0767c5	3 years ago
Kevin Harwell	30cefc97a6	deprecation cleanup: remove leftover files Several modules removal and deprecations occurred in 19.0.0 (initial 19 release), but associated UPGRADE files were not removed from staging for some reason in the master branch. This patch removes those files, and also removes a spurious leftover header, chan_phone.h (associated module removed in 19). Change-Id: Ib92142c846b45c882d6b2b6caca7225253c83add	3 years ago
Joshua C. Colp	fa0078fbe4	pjproject: Update bundled to 2.12 release. This change removes patches which have been merged into upstream and updates some existing ones. It also adds some additional config_site.h changes to restore previous behavior, as well as a patch to allow multiple Authorization headers. There seems to be some confusion or disagreement on language in RFC 8760 in regards to whether multiple Authorization headers are supported. The RFC implies it is allowed, as does some past sipcore discussion. There is also the catch all of "local policy" to allow it. In the case of Asterisk we allow it. ASTERISK-29351 Change-Id: Id39ece02dedb7b9f739e0e37ea47d76854af7191	3 years ago
Naveen Albert	a7cf3979ec	pbx.c: Warn if there are too many includes in a context. The PBX core uses the stack when it comes to includes, which means that a context can only contain strictly fewer than AST_PBX_MAX_STACK includes. If this is exceeded, then warnings will be emitted for each number of includes beyond this if searching for an extension in the including context, and if the extension's inclusion is beyond the stack size, it will simply not be found. To address this, we now check if there are too many includes in a context when the dialplan is reloaded so that if there is an issue, the user is aware of at "compile time" as opposed to "run time" only. Secondly, more details are printed out when this message is encountered so it's clear what has happened. ASTERISK-26719 Change-Id: Ia3700452e75a7af3391b3e82ee69f06a669f8958	3 years ago
George Joseph	3e97156fd3	Makefile: Disable XML doc validation make_xml_documentation was being called with the --validate flag set when it shouldn't have been. This was causing build failures if neither xmllint nor xmlstarlet were installed. The correct behavior is to simply print a message that either one of those tools should be installed for validation and continue with the build. ASTERISK-29988 Change-Id: Idc6c44114e7dd3fadae183a4e22f4fdba0b8a645	3 years ago
George Joseph	144b3c5453	make_xml_documentation: Remove usage of get_sourceable_makeopts get_sourceable_makeopts wasn't handling variables with embedded double quotes in them very well. One example was the DOWNLOAD variable when curl was being used instead of wget. Rather than trying to fix get_sourceable_makeopts, it's just been removed. ASTERISK-29986 Reported by: Stefan Ruijsenaars Change-Id: Idf2a90902228c2558daa5be7a4f8327556099cd2	3 years ago
Naveen Albert	0d11938e92	chan_iax2: Fix spacing in netstats command The iax2 show netstats command previously didn't contain enough spacing in the header to properly align the table header with the table body. This caused column headers to not align with the values on longer channel names. Some spacing is added to account for the longest channel names that display (before truncation occurs) so that columns are always properly aligned. ASTERISK-29895 #close patches: 61205_misaligned2.patch submitted by Birger Harzenetter (license 5870) Change-Id: I450ce6bb81157b9d6d149007e53b749f237b6d9f	3 years ago
Sean Bright	5ac5c2b0ab	openssl: Supress deprecation warnings from OpenSSL 3.0 There is work going on to update our OpenSSL usage to avoid the deprecated functions but in the meantime make it possible to compile in devmode. Change-Id: Ib082eb8b3751f0185d8aa8fe127da664c93f0726	3 years ago
Marcel Wagner	9b654d4e98	documentation: Add information on running install_prereq script in readme Adding information in the readme about running the install_preqreq script to install components that the ./configure script might indicate as missing. ASTERISK-29976 #close Change-Id: Ic287b46300168729838bddd8f9265e98fc22bce6	3 years ago
Naveen Albert	7bc8ef2681	chan_iax2: Fix perceived showing host address. ASTERISK_22025 introduced a regression that shows the host IP and port as the perceived IP and port again, as opposed to showing the actual perceived address. This fixes this by showing the correct information. ASTERISK-29048 #close Change-Id: I0ad3e25bc6b449e83ce72ea5d1a1cdba72aa304a	3 years ago
Boris P. Korzun	6624e34580	res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity Change RTP timer behavior for detecting RTP only after two-way SDP channel establishment. Ignore detecting after receiving 183 with SDP or while direct media is used. Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout and rtpholdtimeout options in chan_sip. ASTERISK-26689 #close ASTERISK-29929 #close Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb	3 years ago
Hugh McMaster	64f11e0d18	configure.ac: Use pkg-config to detect libxml2 Use pkg-config to detect libxml2, falling back to xml2-config if the former is not available. This patch ensures Asterisk continues to build on systems without xml2-config installed. The patch also updates the associated 'configure' files. ASTERISK-29970 #close Change-Id: I3c90dfe0b0590486cbb8e6d426a7c5c4199410c0	3 years ago
Philip Prindeville	287a1a9126	time: add support for time64 libcs Treat time_t's as entirely unique and use the POSIX API's for converting to/from strings. Lastly, a 64-bit integer formats as 20 digits at most in base10. Don't need to have any 100 byte buffers to hold that. ASTERISK-29674 #close Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com> Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56	3 years ago
Alexei Gradinari	d1900d4a4c	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	3 years ago
Naveen Albert	1e87cadf8e	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	3 years ago
Sean Bright	d3abdf0b8d	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	3 years ago
Sean Bright	686c386b05	download_externals: Use HTTPS for downloads ASTERISK-29980 #close Change-Id: I7b347665822ea2774dd322276c09be67914d2065	3 years ago
Sean Bright	c33718a54d	conversions.c: Specify that we only want to parse decimal numbers. Passing 0 as the last argument to strtoimax() or strtoumax() causes octal and hexadecimal to be accepted which was not originally intended. So we now force to only accept decimal. ASTERISK-29950 #close Change-Id: I93baf0f273441e8280354630a463df263a8c0edd	3 years ago
Philip Prindeville	2a87303ebd	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	3 years ago
Naveen Albert	fd29d28832	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	3 years ago
Alexei Gradinari	edce853123	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	3 years ago
Naveen Albert	37ece75677	res_agi: Fix xmldocs bug with set music. The XML documentation for the SET MUSIC AGI command is invalid, as the parameter does not have a name and the on/off enum options for the on/off argument are listed separately, which is incorrect. The cumulative effect of these currently is that the Asterisk Wiki documentation for SET MUSIC is broken and external documentation generators crash on SET MUSIC due to the malformed documentation. These issues are corrected so that the documentation can be successfully parsed as with other similar AGI commands. ASTERISK-29939 #close ASTERISK-28891 #close Change-Id: I8c3d59897531bcbc401cbc7b00c9e2829dcb35f8	3 years ago
Boris P. Korzun	636d43caa3	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	3 years ago
Kfir Itzhak	2be01ba40b	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	3 years ago
Naveen Albert	fbde0186c7	ami: Improve substring parsing for disabled events. ASTERISK_29853 added the ability to selectively disable AMI events on a global basis, but the logic for this uses strstr which means that events with names which are the prefix of another event, if disabled, could disable those events as well. Instead, we account for this possibility to prevent this undesired behavior from occuring. ASTERISK_29853 Change-Id: Icccd1872602889806740971e4adf932f92466959	3 years ago
George Joseph	b40c4d59b1	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	3 years ago
George Joseph	9c36c055c1	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	3 years ago
George Joseph	b5391ff691	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	3 years ago
George Joseph	2e00b5edbd	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	3 years ago
George Joseph	1950cec3fd	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	3 years ago
Joshua C. Colp	dd7db5c698	jansson: Update bundled to 2.14 version. ASTERISK-29353 Change-Id: I4ea43eda1691565563a4c03ef37166952d211b2b	3 years ago
Naveen Albert	27fb4fd5bc	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	3 years ago
Naveen Albert	3a3b8fbd9f	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	3 years ago
Naveen Albert	2ba5da15b0	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	3 years ago
Naveen Albert	c35e205bef	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	3 years ago

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