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2009-12-10Add audio announcement option to app_pagejpeeler1-2/+32
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: http://svn.digium.com/svn/asterisk/trunk@234173 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-15Last batch of 'static' qualifiers for module-level global variables.kpfleming1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@200656 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Const-ify the world (or at least a good part of it)kpfleming1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-12add 'const' qualifiers in various places where they should have beenkpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@193832 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-25Merged revisions 170979 via svnmerge from seanbright1-4/+7
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: http://svn.digium.com/svn/asterisk/trunk@170980 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-14Merged revisions 168608 via svnmerge from murf1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@168613 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-14Merged revisions 168593 via svnmerge from twilson1-4/+16
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: http://svn.digium.com/svn/asterisk/trunk@168594 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Add an 'i' option to app_page. This option works the same asmmichelson1-0/+9
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: http://svn.digium.com/svn/asterisk/trunk@164428 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05- Add more <see-also> based on TFOT.eliel1-0/+3
- Add the 'filename' type to the see-also ref. To be able to reference a filename. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154578 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-04Slightly optimize ast_devstate_str and rename global functions devstate2str ↵tilghman1-2/+2
and config_text_file_save to have an ast_ prefix git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154260 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02fix a typo (thanks sean)russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153470 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02Fix various spelling and grammatical issues in documentationrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153468 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01Merge changes from team/group/appdocsxmlrussell1-17/+50
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: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-31* Fixed timeout logic in the dialing API as setting timeoutsmmichelson1-10/+35
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: http://svn.digium.com/svn/asterisk/trunk@153223 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-12Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵jpeeler1-1/+1
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-11Dial a device even if it's state is unknown.file1-3/+8
(closes issue #12184) Reported by: bluecrow76 Patches: asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by bluecrow76 (license 270) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107710 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-3/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-4/+0
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: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵tilghman1-3/+3
few other formatting cleanups, too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-23Merge the dialplan_aesthetics branch. Most of this patch simply converts ↵tilghman1-3/+3
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: http://svn.digium.com/svn/asterisk/trunk@76703 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-6/+0
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16It is no longer required for each module that deals with a channel to call ↵file1-7/+1
ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-01Merged revisions 66879 via svnmerge from russell1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@66880 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-16Merged revisions 58992 via svnmerge from file1-3/+2
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: http://svn.digium.com/svn/asterisk/trunk@58995 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-12Merged revisions 54066 via svnmerge from russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@54067 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-24Merged revisions 52049 via svnmerge from file1-79/+44
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: http://svn.digium.com/svn/asterisk/trunk@52050 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-18Add 's' option to Page application which checks devicestate before dialing. ↵file1-6/+18
(issue #8673 reported by sunder) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51215 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-20Clean up app_pagefile1-33/+41
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48639 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-26Merged revisions 46347 via svnmerge from qwell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46347 | qwell | 2006-10-26 15:25:44 -0500 (Thu, 26 Oct 2006) | 2 lines Fix small formatting issue, that causes misaligned line ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46348 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-11Merged revisions 42783 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r42783 | tilghman | 2006-09-11 16:47:23 -0500 (Mon, 11 Sep 2006) | 4 lines When paging, only wait 5 seconds for the marked user to enter the conference. After that, assume the paging already completed by the time the channel entered the conference and drop back out. (Issue 7275) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42788 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21merge new_loader_completion branch, including (at least):kpfleming1-19/+8
- restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-4/+4
again :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10remove almost all of the checks of the result from ast_strdupa() or alloca().russell1-4/+1
As it turns out, all of these checks were useless, because alloca will never return NULL. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26451 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-03 Documentation / whitespace fix.bweschke1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@24639 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-03 Optionally record audio of the page command for re-pages/playback. #6827 ↵bweschke1-3/+8
(JeffSaxe) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@24638 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-24Thanks to the fine work of Russell Bryant and Dancho Lazarov, we now have ↵kpfleming1-0/+5
autoconf and menuselect tools for Asterisk! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22267 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-14This rather large commit changes the way modules are loaded. rizzo1-16/+7
As partly documented in loader.c and include/asterisk/module.h, modules are now expected to return all of their methods and flags into a structure 'mod_data', and are normally loaded with RTLD_NOW | RTLD_LOCAL, so symbols are resolved immediately and conflicts should be less likely. Only in a small number of cases (res_*, typically) modules are loaded RTLD_GLOBAL, so they can export symbols. The core of the change is only the two files loader.c and include/asterisk/module.h, all the rest is simply adaptation of the existing modules to the new API, a rather mechanical (but believe me, time and finger-consuming!) process whose detail you can figure out by svn diff'ing any single module. Expect some minor compilation issue after this change, please report it on mantis http://bugs.digium.com/view.php?id=6968 so we collect all the feedback in one place. I am just sorry that this change missed SVN version number 20000! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20003 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-13Merged revisions 19812 via svnmerge from kpfleming1-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r19812 | kpfleming | 2006-04-13 12:40:21 -0500 (Thu, 13 Apr 2006) | 2 lines oops... let's not set a variable and then immediately overwrite it while assuming its old value will magically return ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@19813 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-11Merged revisions 19348 via svnmerge from kpfleming1-1/+12
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r19348 | kpfleming | 2006-04-11 16:50:18 -0500 (Tue, 11 Apr 2006) | 2 lines don't call the originating device as part of the Page() operation (issue #6932) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@19350 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-08since the module API is changing, it's a good time to const-ify the ↵kpfleming1-2/+2
description() and key() return values git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18552 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-05Bug 6873 - Finish moving from the non-threadsafe (and poor randomness) ↵tilghman1-1/+1
rand() to threadsafe ast_random() git-svn-id: http://svn.digium.com/svn/asterisk/trunk@17627 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-30as discussed with Mark a few weeks ago, the 'newstack' argumentrizzo1-1/+1
in pbx_exec is always 1 so it can be removed. This change also takes away ast_exec_extension(), and lets all switch functions (exists, canmatch, exec, matchmore) all use the same prototype, which makes the code a bit cleaner. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16558 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15remove the uses of the deprecated STANDARD_LOCAL_USERrussell1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10241 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-11update for change to ast_pbx_outgoing_extenrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9601 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-21on this pass, only remove duplicate log messagesrussell1-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8403 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-21revert my pass through the tree to remove checks of the result of ast_strduparussell1-0/+5
(revisions 8378 through 8381) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8387 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-21remove lots of useless checks of the result of ast_strduparussell1-5/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8379 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-13 More memory wrapper cleanup. #6224bweschke1-3/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8045 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-12 More new memory wrapper work.bweschke1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8012 f38db490-d61c-443f-a65b-d21fe96a405b
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