87 Commits (master)

Author	SHA1	Message	Date
Sean Bright	82bef2ed90	docs: Indent <since> tags. ... Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0).	3 months ago
George Joseph	198b01c536	docs: Add version information to application and function XML elements ... * Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml	3 months ago
Alexander Traud	196c24df22	apps: Fix for Doxygen. ... ASTERISK-29740 Change-Id: Icb6fbcfea0a5f1c82caa5001902b6a786adbf307	3 years ago
Sean Bright	932eae69ab	app_page.c: Don't fail to Page if beep sound file is missing ... ASTERISK-16799 #close Change-Id: I40367b0d6dbf66a39721bde060c8b2d734a61cf4	4 years ago
Richard Mudgett	d86a6ac5ce	app_page.c: Simplify dialplan using Page. ... Dialplan has to be careful about passing an empty destination list or empty positions in the list. As a result, dialplan has to check for these conditions before using Page. Simplify dialplan by making Page handle these conditions gracefully. * Made tolerate empty positions in the paged device list. * Reduced some warnings associated with the 's' option to verbose messages. The warning level for those messages really serves no purpose as that is why the 's' option exists. ASTERISK-28638 Change-Id: I95b64a6d6800cd1a25279c88889314ae60fc21e3	5 years ago
Corey Farrell	52b530503f	app_page: Add dependency against app_confbridge. ... Change-Id: I1946509f518961d23fb21229d91676ee3e441921	7 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	7 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Walter Doekes	39daf9f066	docs: Fix a few typo's in app docs (more then, resourse). ... Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7	10 years ago
Richard Mudgett	06f4f80a63	app_page.c: Fix crash when forwarding with a predial handler. ... Page uses the async method of dialing with the dial API. When a call gets forwarded there is no calling channel available. If the predial handler was set then the calling channel could not be put into auto-service for the forwarded call because it doesn't exist. A crash is the result. * Moved the callee predial parameter string processing to before the string is passed to the dial API rather than having the dial API do it. There are a few benefits do doing this. The first is the predial parameter string processing doesn't need to be done for each channel called by the dial API. The second is in async mode and the forwarded channel is to have the predial handler executed on it then the non-existent calling channel does not need to be present to process the predial parameter string. * Don't start auto-service on a non-existent calling channel to execute the predial handler when the dial API is in async mode and forwarding a call. ASTERISK-25384 #close Reported by: Chet Stevens Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981	10 years ago
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	10 years ago
Scott Griepentrog	80ef9a21b9	uniqueid: channel linkedid, ami, ari object creation with id's ... Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	f6e92c35df	app_page: Add predial handlers for app_page. ... (closes issue AFS-14) Review: https://reviewboard.asterisk.org/r/3045/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	fe6fc6e3b0	app_page and app_confbridge: Fix custom announcement on entering conference. ... The Page and ConfBridge custom announcement did not play when users entered the conference. * Fix the CONFBRIDGE(user,announcement) file not getting played. The code to do this got removed accidentally when the ConfBridge code was restructured to be more state machine like. * Fixed play_prompt_to_user() doxygen comments. * Fixed the Page A(x) and n options for the caller. The caller never played the announcement file and totally ignored the n option. The code to do this was lost when the application was converted to use ConfBridge. * Factored out setup_profile_bridge(), setup_profile_paged(), and setup_profile_caller() routines to setup ConfBridge profiles. Made each profile setup routine use the default template if one has not already been setup by dialplan. (closes issue ASTERISK-20990) Reported by: Jeremy Kister Tested by: rmudgett ........ Merged revisions 380894 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	fb323d4465	app_page: Fixup application XML documentation typos and inaccuracies. ... ........ Merged revisions 380869 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7b51320642	Fix a variety of memory leaks ... This patch addresses a number of memory leaks in a variety of modules that were found by a static analysis tool. A brief summary of the changes: * app_minivm: free ast_str objects on off nominal paths * app_page: free the ast_dial object if the requested channel technology cannot be appended to the dialing structure * app_queue: if a penalty rule failed to match any existing rule list names, the created rule would not be inserted and its memory would be leaked * app_read: dispose of the created silence detector in the presence of off nominal circumstances * app_voicemail: dispose of an allocated unique ID field for MWI event un-subscribe requests in off nominal paths; dispose of configuration objects when using the secret.conf option * chan_dahdi: dispose of the allocated frame produced by ast_dsp_process * chan_iax2: properly unref peer in CLI command "iax2 unregister" * chan_sip: dispose of the allocated frame produced by sip_rtp_read's call of ast_dsp_process; free memory in parse unit tests * func_dialgroup: properly deref ao2 object grhead in nominal path of dialgroup_read * func_odbc: free resultset in off nominal paths of odbc_read * cli: free match_list in off nominal paths of CLI match completion * config: free comment_buffer/list_buffer when configuration file load is unchanged; free the same buffers any time they were created and config files were processed * data: free XML nodes in various places * enum: free context buffer in off nominal paths * features: free ast_call_feature in off nominal paths of applicationmap config processing * netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct that is allocated by the method. Failures in ast_sockaddr_resolve could result in the users of the method not knowing whether or not the buffer was allocated. The method will now not allocate the ast_sockaddr struct if it will return failure. * pbx: cleanup hash table traversals in off nominal paths; free ignore pattern buffer if it already exists for the specified context * xmldoc: cleanup various nodes when we no longer need them * main/editline: various cleanup of pointers not being freed before being assigned to other memory, cleanup along off nominal paths * menuselect/mxml: cleanup of value buffer for an attribute when that attribute did not specify a value * res_calendar*: responses are allocated via the various *_request method returns and should not be allocated in the various write_event methods; ensure attendee buffer is freed if no data exists in the parsed node; ensure that calendar objects are de-ref'd appropriately * res_jabber: free buffer in off nominal path * res_musiconhold: close the DIR* object in off nominal paths * res_rtp_asterisk: if we run out of ports, close the rtp socket object and free the rtp object * res_srtp: if we fail to create the session in libsrtp, destroy the temporary ast_srtp object (issue ASTERISK-19665) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1922 ........ Merged revisions 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 366881 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	0ec73946fa	app_page: Fix a memory leak on every Page(). ... dial_list is a dynamically allocated array that is allocated at the beginning of Page() based on how many devices will be dialed. This was never being freed. ........ Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 360364 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f5fda0eb74	Transition app_page to using app_confbridge internally for the conference bridge portion of paging. This also adds a new 'announcement' option to ConfBridge user profiles. ... Review: https://reviewboard.asterisk.org/r/1754/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	39ffc96723	Remove dupliate 'i' option table entry in app_page.c. ... (closes issue ASTERISK-19310) Reported by: Makoto Dei Patches: app_page-duplicate-i-option.patch (license #5027) patch uploaded by Makoto Dei ........ Merged revisions 357352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 357353 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields ... Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions ... There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jeff Peeler	2414bc8005	Add audio announcement option to app_page ... As described in the CHANGES file: * MeetMe has a new option 'G' to play an announcement before joining a conference. * Page has a new option 'A(x)' which will playback an announcement simultaneously to all paged phones (and optionally excluding the caller's one using the new option 'n') before the call is bridged. To add the new option to meetme, the conference flag options had to be extended to 64 bits. (closes issue #14365) Reported by: dferrer Patches: page_announce.patch uploaded by dferrer (license 525) modified by me Review: https://reviewboard.asterisk.org/r/188/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	aaeec3b40f	Last batch of 'static' qualifiers for module-level global variables. ... Fix up modules in the 'apps' directory, and also correct the bad example of enum definitions in include/asterisk/app.h, which many developers followed (thanks for reading the documentation!). In addition, add some basic usage examples of the 'pahole' and 'pglobal' tools to the coding guidelines. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) ... This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	1c988d8996	add 'const' qualifiers in various places where they should have been ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	ec5aa59105	Merged revisions 170979 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Steve Murphy	a474341105	Merged revisions 168608 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	2015f9c980	Merged revisions 168593 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines Don't overflow when paging more than 128 extensions The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	763d4dcabb	Add an 'i' option to app_page. This option works the same as ... the 'i' options for app_dial and app_queue, in that they will ignore any attempts by phones to forward the call. (closes issue #13977) Reported by: putnopvut Patches: page_ignore_forwards.patch uploaded by putnopvut (license 60) Tested by: putnopvut, acunningham git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164428 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	f18699be24	- Add more <see-also> based on TFOT. ... - Add the 'filename' type to the see-also ref. To be able to reference a filename. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	2cc8e25222	Slightly optimize ast_devstate_str and rename global functions devstate2str and config_text_file_save to have an ast_ prefix ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2baafa6156	fix a typo (thanks sean) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	6f314f4d42	Fix various spelling and grammatical issues in documentation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b168ee34b	Merge changes from team/group/appdocsxml ... This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	d521ad9696	* Fixed timeout logic in the dialing API as setting timeouts ... had no effect * Updated dialing API documentation to indicate that timeouts are specified in milliseconds * Added a new timeout argument to the Page application. If time expires, any endpoints which have not answered will be hung up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	ef3b214728	Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	b9a6dbf587	Dial a device even if it's state is unknown. ... (closes issue #12184) Reported by: bluecrow76 Patches: asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by bluecrow76 (license 270) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Luigi Rizzo	7e8835e0d7	remove another set of redundant #include "asterisk/options.h" ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	0595b5e2aa	include "logger.h" and errno.h from asterisk.h - usage shows that they ... were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect ... build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	20bbd09de3	Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	55b1ee298e	Merge the dialplan_aesthetics branch. Most of this patch simply converts applications ... using old methods of parsing arguments to using the standard macros. However, the big change is that the really old way of specifying application and arguments separated by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76703 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	b8cd949cce	Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	96a646734f	It is no longer required for each module that deals with a channel to call ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	5af2d0720e	Merged revisions 66879 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r66879 | russell | 2007-06-01 14:35:13 -0500 (Fri, 01 Jun 2007) | 2 lines List app_meetme as a module that app_page depends on. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	bbcbf46aae	Merged revisions 58992 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58992 | file | 2007-03-16 12:12:28 -0400 (Fri, 16 Mar 2007) | 2 lines Wait for the async thread to exit when hanging up all of the paged phones under all circumstances. (issue #9181 reported by PhilSmith) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58995 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	2a5477b35e	Merged revisions 54066 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines - Add the ability to register a callback to monitor state changes in an asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	9826fc599b	Merged revisions 52049 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2 lines Merge in dialing API and the app_page that uses it. (issue #BE-118) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Joshua Colp	dd23f68d18	Add 's' option to Page application which checks devicestate before dialing. (issue #8673 reported by sunder) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago