| Age | Commit message (Collapse) | Author |
|---|
|
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0
|
|
Use -Wno-format-truncation only if supported by compiler.
ASTERISK-27171 #close
Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6
|
|
GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
ASTERISK-27156 #close
Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
|
|
This patch updates a variety of Makefiles in Asterisk's build system to
remove .gcda and .gcno files when 'make clean' is executed. These files
are generated when '--enable-coverage' is passed to the Asterisk
configure script.
Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602
|
|
This new macro allows a single line to add all additional
sources to a module. This helps prevent modules from
missing steps, and makes future changes easier since
they can be made in a single place.
ASTERISK-24960 #close
Reported by: Corey Farrell
Change-Id: I38f12d8b72c5e7bb37a879b2fb51761a2855eb4b
|
|
Both the apps and channels Makefiles still listed 'res_features' as modules to
link against when compiling for cygwin or mingw32. This module hasn't existed
for quite some time.
ASTERISK-18105 #close
Reported by: feyfre
........
Merged revisions 432341 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 432342 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.
Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.
The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.
For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.
And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.
To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.
Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.
We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.
It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.
And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.
Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.
This patch removes:
* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge
It removes the following applications/functions:
* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO
It removes the colon delimiter from the SIPPEER function.
Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.
Review: https://reviewboard.asterisk.org/r/3698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 403736 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403737 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Review: https://reviewboard.asterisk.org/r/3038/
........
Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
........
Merged revisions 397643 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Created a native_dahdi bridging technology for use with the new bridging
API.
The new bridging technology is part of the chan_dahdi channel driver
because it is very specific to that driver. Rather than include the new
code directly into chan_dahdi.c the new bridge technology is in its own
file and linked into chan_dahdi.so. A large part of this change is the
mechanical process of moving declarations around so chan_dahdi.c can be
split up into more files later.
* Changed the bridging core to pass NULL frames into the channel
technologies instead of discarding them. The channel technologies may
need the proding to determine if their configuration is still valid.
(closes issue ASTERISK-21886)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2681/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.
SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.
API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This patch just moves the IAX2 source and header files into a separate iax2
sub-directory in the channels directory, similar to how the sip source files are
structured.
The only thing that was added was an #ifndef to protect provision.h from multiple
inclusion.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Update and extend the configuration_file group and enable linking. Update title that was left behind many years ago.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Extract the SS7 specific code out of chan_dahdi like what was done to
ISDN/PRI and analog signaling. The new SS7 structures were modeled on
sig_pri.
The changes to sig_pri are an enhancement and a bug fix made possible
because SS7 was extracted.
1) The sig_pri TRANSFERCAPABILITY channel variable should have been set
unconditionally in sig_pri_new_ast_channel().
2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of
SS7 extraction.
3) Module ref count error in dahdi_new() if startpbx failed to start the
PBX for some reason.
Review: https://reviewboard.asterisk.org/r/661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
After some debugging, the random chan_h323 build failures appear to be due
to complications introduced by some chan_h323 specific build stuff getting
triggered during a clean. Simplify this by moving the h323 clean commands
down into channels/makefile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
users expect them to work.
'core set debug' and 'core set verbose' can optionally change the
level for a specific filename; however, this is actually for a
specific source file name, not the module that source file is included
in. With examples like chan_sip, chan_iax2, chan_misdn and others
consisting of multiple source files, this will not lead to the
behavior that users expect. If they want to set the debug level for
chan_sip, they want it set for all of chan_sip, and not to have to
also set it for reqresp_parser and other files that comprise the
chan_sip module.
This patch changes this functionality to be module-name based instead
of file-name based.
To make this work, some Makefile modifications were required to ensure
that the AST_MODULE definition is present in each object file produced
for each module as well.
Review: https://reviewboard.asterisk.org/r/574/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
when we
change flags. Fixes a weird symbol problem mmichelson was having in a group branch,
but also applies to trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248667 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
CHANNEL function.
Also includes a test for retrieving rtpqos parameters, including a NULL RTP
driver. Additionally, some further separation of the SIP internal API into
headers was necessary.
(closes issue #16652)
Reported by: kkm
Patches:
20100204__issue16652.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/501/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247124 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
They were previously passed correctly, but they simply weren't used. This
caused issues with various platforms whose builds needed to pass special
linker flags via the configure script.
(closes issue #16596)
Reported by: pprindeville
Patches:
asterisk-1.6-astldflags.patch uploaded by pprindeville (license 347)
Tested by: tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
New files
- channels/sip/sip.h – A new header for shared #define, enum, and struct
definitions.
- channels/sip/include/sip_utils.h – sip util functions shared among
the all the sip APIs
- channels/sip/include/config_parser.h – sip config-parser API
- channels/sip/config_parser.c – Contains sip.conf parsing helper functions
with unit tests.
- channels/sip/include/reqresp_parser.h – sip request response parser API
- channels/sip/reqresp_parser.c – Contains sip request and response parsing
helper functions with unit tests.
New Unit Tests
- sip_parse_uri_test
- sip_parse_host_test
- sip_parse_register_line_test
Code Refactoring
- All reusable #define, enum, and struct definitions were moved out of chan_sip.c
into sip.h. During this process formatting changes were made to comments
in both sip.h and chan_sip.c in order to better adhere to the coding guidelines.
- The beginnings of three new sip APIs, sip-utils.h, config-parser.h,
reqresp-parser.h using existing chan_sip.c functions.
- parse_uri() and get_calleridname() were moved from chan_sip.c to request-parser.c
along with unit tests for both functions.
- sip_parse_host() and sip_parse_register_line() were moved from chan_sip.c to
config-parser.c along with unit tests for both functions.
Changes to parse_uri()
-removal of the options parameter. It was never used and did not behave correctly.
-additional check for [?header] field. When this field was present, the transport
type was not being set correctly.
----- Overview -----
This patch is introduced with the hope that unit tests for all our sip parsing
functions will be written soon. chan_sip is a huge file, and with the addition of
each unit test chan_sip is going to grow larger and harder to maintain. I'm proposing
we begin refactoring chan_sip, starting with the parsing functions. With each parsing
function we move into a separate helper file, a unit test should accompany it. I've
attempted to lay down the ground work for this change by creating two new parser
helper files (config-parser.c and reqresp-parser.c) and moving all shared structs,
enums, and defines from chan_sip.c into a shared sip.h file. We can't verify everything
in Asterisk using unit tests, but string parsing is one area where unit tests make
the most sense. By beginning to restructure the code in this way, chan_sip not only
becomes less bloated, but Asterisk as a whole will become more stable.
Review: https://reviewboard.asterisk.org/r/477/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
Ensure that user-provided CFLAGS and LDFLAGS are honored.
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@207680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into
sig_pri.c. Functionality in theory should not change (mostly). A few trivial
changes were made in sig_analog with verbose messages and commenting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
This branch splits all the analog signaling logic out of chan_dahdi.c into
sig_analog.c. Functionality in theory should not change at all. As noted
in the code, there is still some unused code remaining that will be cleaned
up in a later commit.
Review: https://reviewboard.asterisk.org/r/253/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Include libchanh323.a in the modules.link file so that all the symbols can be
resolved at link time.
(closes issue #11966)
Reported by: dome
Patches:
issue_11966.patch uploaded by kpfleming (license 421)
Tested by: jpeeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(closes issue #14715)
Reported by: jthurman
Patches:
h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614)
Tested by: tzafrir, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
GCC 4.3.2 to complain that the passed argument will always evaluate to
true. So to get things to compile, disable assert when building
chan_usbradio.so.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar 2008) | 2 lines
get chan_vpb to build properly in dev mode
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
The main code to implement the textarea is in console_board.c,
and uses a simple png image with the font, blitting characters
on the designated areas of the main screen.
Additionally we provide some annotations in the image used
as a skin to indicate which areas are used for text messages.
(images will be committed separately).
At the moment the dialog area is only used to display a running
counter, just as a proof of concept.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
indications can handle it
remove gentone and all the headers containing tones that are no longer needed
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
to add more entries. This required moving struct grab_desc to the common
header, and adding an entry in the Makefile.
On passing, cleanup some comments and file headers (some are still missing).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
do not change.
Ths masks (but does not solve) a but that i am seeing in doing a
'gmake install' without donig a 'gmake all' first.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
console_video.c
This will ease the task of splitting console_video.c into its components
(V4L and X11 grabbers, various video codecs and packetizers, SDL),
as well as ease future extensions (e.g. additional video sources,
codecs and rendering engines).
For the time being nothing changes for users: video support is off by
default, and requires -DHAVE_VIDEO_CONSOLE on the command line to be included
(if SDL and FFMPEG are available).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
and report that information so that menuselect can use it (all information that is used to decide whether to build modules or not must be fed to menuselect so the user knows what will be built and why... don't make module build decisions in the makefiles, please)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines
In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
rizzo brought up some issues related to the way that the metadata required
for menuselect and the rest of the build system is extracted from the source
files. Since I had a few hours to kill on an airplane today, I decided to
improve this situation... so now the system caches the extracted metadata
and uses it to build the menuselect 'tree' as much as it can. The result
of this is that when a single source file is changed, only the metadata for
that file needs to be extracted again, and the rest is used from the cache
files. I also reduced the number of forked processes required to do the
metadata extraction; it was actually possible to do most of what we needed
in the Makefiles themselves without using any shell scripts at all! On my
laptop, these changes resulted in an 80% decrease in the time required
for the 'menuselect.makeopts' automatic check to occur after editing a single
source file.
While doing this work I also cleaned up a few minor things in the Makefiles,
adding a check for 'awk' to the configure script and changed all remaining
places we use 'grep' or 'awk' to use the ones found by the configure script,
and changed the 'prep_tarball' script to build the menuselect metadata so
that tarballs of Asterisk will include it and won't require the user to
wait while it is extracted after unpacking.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
generate loadable and embedded module lists.
Individual Makefiles now are a lot simpler, possibly as simple as this:
-include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
MODULE_PREFIX=cdr_
all: _all
include $(ASTTOPDIR)/Makefile.moddir_rules
and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.
The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).
With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
the top level directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
After this commit we can actually load modules under windows,
and we can start debugging more interesting problems related
to the load order and functionality of modules.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
(this should be handled with the dependencies generated by
configure and menuselect, but will be fixed later)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89366 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
Remove unset variables, they would be misleading.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1 line
Only try to clean out h323/ if the h323/Makefile exists.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r84291 | qwell | 2007-10-01 16:52:45 -0500 (Mon, 01 Oct 2007) | 6 lines
Add dist-clean support for subdirs.
Change h323 to only remove the Makefile on a dist-clean, rather than a clean.
This fixes a bug I found with trying to run make after a make clean
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
|
from IgorG.
Merged revisions 84170 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r84170 | russell | 2007-10-01 10:00:56 -0500 (Mon, 01 Oct 2007) | 3 lines
Remove another file in "make clean".
(closes issue #10814, paravoid)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
