Age | Commit message (Collapse) | Author |
|
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
|
|
........
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
|
|
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
|
|
Review: https://reviewboard.asterisk.org/r/1770/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Review: https://reviewboard.asterisk.org/r/1753/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
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
|
|
Reviewboard: https://reviewboard.asterisk.org/r/416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|
|
(closes issue #14341)
Reported by: fnordian
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
<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
|
|
Patch by me (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|
|
app_voicemail and app_queue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(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
|
|
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
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89518 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|
|
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
|
|
seanbright while I was debugging issue 11109.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
few other formatting cleanups, too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|
|
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
|
|
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
|
|
(Closes issue 10061, Reported and patched by eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(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
|
|
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
|
|
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
|
|
compile under --enable-dev-mode
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|
|
patch provided in bugnote, with minor changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@42715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|
|
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
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
- 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
|
|
again :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
julien23)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@29435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
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
|