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2013-12-05app_record: Add an option that allows DTMF '0' to act as an additional ↵Jonathan Rose
terminator Using this terminator when active results in ${RECORD_STATUS} being set to 'OPERATOR' instead of 'DTMF' (closes issue AFS-7) Review: https://reviewboard.asterisk.org/r/3041/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-07Multiple revisions 375993-375994Mark Michelson
........ r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) | 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 | mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-18Fix a variety of memory leaksMatthew Jordan
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
2012-02-24Opaquification for ast_format structs in struct ast_channelTerry Wilson
Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-20ast_channel opaquification of pointers and integral typesTerry Wilson
Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-24Opaquify channel stringfieldsTerry Wilson
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
2012-01-09Replace direct access to channel name with accessor functionsTerry Wilson
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
2011-07-14Merged revisions 328247 via svnmerge from Leif Madsen
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
2011-02-03Asterisk media architecture conversion - no more format bitfieldsDavid Vossel
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-10Be less ambiguous in Record() app docs.Leif Madsen
For some reason the documentation for the 'k' application in trunk and 1.6.2 is different than 1.6.0 and 1.6.1, so I'm setting them all to match. The wording in 1.6.2 and trunk was ambiguous, so you could interpret the wording the mean that recording would continue upon hangup indefinitely, or you could interpret it to mean that the recorded data would not be discarded upon hangup. This change makes it clear we mean the latter, and not the former. Came from a discussion in #asterisk on IRC. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-02Add an option to Record which enables a mode where any DTMF digit will ↵Joshua Colp
terminate recording. (closes issue #15436) Reported by: Vince Patches: app_record.diff uploaded by Vince (license 823) Tested by: dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Expand codec bitfield from 32 bits to 64 bits.Tilghman Lesher
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10AST-2009-005Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
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
2009-02-13Add an option to keep the recorded file upon hangup.Joshua Colp
(closes issue #14341) Reported by: fnordian git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19Fix the XML documentation for Record().Eliel C. Sardanons
<value> tags inside <variable> elements must have CDATA and no another XML node. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Add RECORD_STATUS variable, as requested on the -users list.Tilghman Lesher
Patch by me (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16(closes issue #13669)Jeff Peeler
Reported by: pj Delete file recording if recording terminated from a hangup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02Another little one.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02Fix various spelling and grammatical issues in documentationRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01Merge changes from team/group/appdocsxmlRussell Bryant
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
2008-08-10More RSW merges. Everything from apps/ except for the big offendersSean Bright
app_voicemail and app_queue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05Create a centralized configuration option for silencethresholdTilghman Lesher
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-22shuffle a little bit the content of header files to reduce dependencies.Luigi Rizzo
In this commit: - move the ast_register/unregister_app functions to module.h to avoid the need to include pbx.h for the simpler apps; - move the ast_group structure to channel.h to remove the dependency of app.h on linkedlists.h Note, this is a long process that I am doing in small steps. The main difficulty is that now for each subsystem we have a single header (e.g. channel.h) included by the subsystem provider (usually one file, e.g. channel.c) and by its clients (dozens of them, e.g. we have some 70+ apps and 30+ functions). This requires the clients to include all the extra headers required by the provider (eg. lock.h, linkedlists.h, definitions of substructures...) even though many of the clients would be just happy with opaque struct declarations and function prototypes. The long term plan is to eventually rectify this structure so that the compilation can become faster, and also APIs are more stable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-22remove redundant headersLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21remove another set of redundant #include "asterisk/options.h"Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyLuigi Rizzo
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
2007-11-16Start untangling header inclusion in a way that does not affectLuigi Rizzo
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
2007-10-29Make sure we set flags to a 0 value before trying to use it. Pointed out by ↵Jason Parker
seanbright while I was debugging issue 11109. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵Tilghman Lesher
few other formatting cleanups, too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-23Merge the dialplan_aesthetics branch. Most of this patch simply converts ↵Tilghman Lesher
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
2007-07-16Applications no longer need to call ast_module_user_add and ↵Joshua Colp
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
2007-07-16It is no longer required for each module that deals with a channel to call ↵Joshua Colp
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
2007-06-26Create directory if it does not exist.Mark Michelson
(Closes issue 10061, Reported and patched by eliel) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.Russell Bryant
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69327 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-26Merged revisions 56839 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56839 | file | 2007-02-26 14:33:48 -0500 (Mon, 26 Feb 2007) | 2 lines Update app_record documentation to use new CLI command, core show file formats. (issue #9151 reported by junky) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-18Merged revisions 55278 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r55278 | oej | 2007-02-18 13:35:54 +0100 (Sun, 18 Feb 2007) | 10 lines Merged revisions 55277 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 lines Documentation update (#9053, jsmith) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-21a quick fix to app_sms.c to get rid of cursed compiler warnings so I can ↵Steve Murphy
compile under --enable-dev-mode git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-25Merged revisions 46200 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006) | 2 lines apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-10-03bug #8076 check option_debug before printing to debug channel.Matt O'Gorman
patch provided in bugnote, with minor changes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-11small fix for issue 007926Anthony LaMantia
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-09In trunk, ast_fileexists now returns 0 for failure, instead of -1, as it didTilghman Lesher
in 1.2. This corrected a longstanding confusion about the return value. Unfortunately, it broke this app in the process. (Issue 7906) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-09-07Merged revisions 42355 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r42355 | tilghman | 2006-09-07 18:12:29 -0500 (Thu, 07 Sep 2006) | 2 lines Format vulnerability fix - allowing the user to specify a format is not a good idea (Bug 7811) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31spacing updateAnthony LaMantia
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31small typoAnthony LaMantia
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-31fix for bugid: 0007830Anthony LaMantia
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-21merge new_loader_completion branch, including (at least):Kevin P. Fleming
- 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: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵Kevin P. Fleming
again :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-22don't stop recording until hangup (no terminator) (issue #7168 reported by ↵Joshua Colp
julien23) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-05-10remove almost all of the checks of the result from ast_strdupa() or alloca().Russell Bryant
As it turns out, all of these checks were useless, because alloca will never return NULL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-03 02:00:07 +0000