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r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) | 10 lines
Merged revisions 112711 via svnmerge from
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r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 lines
Pass in the path to Zaptel for systems that install Zaptel headers in a separate location.
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r109229 | twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines
Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory.
This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros.
If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect.
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r107409 | kpfleming | 2008-03-11 09:09:49 -0500 (Tue, 11 Mar 2008) | 13 lines
Merged revisions 107408 via svnmerge from
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r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar 2008) | 5 lines
check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4)
(closes issue #12179)
Reported by: Netview
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r107373 | kpfleming | 2008-03-11 06:36:51 -0500 (Tue, 11 Mar 2008) | 19 lines
Merged revisions 107352 via svnmerge from
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r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines
fix up various compiler warnings found with gcc-4.3:
- the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function)
- main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement
- main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur
- main/editline/readline.c had an unused variable
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r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008) | 7 lines
Compatibility fix for PPC64
(closes issue #12081)
Reported by: jcollie
Patches:
asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
Tested by: jcollie, Corydon76
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r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008) | 4 lines
Fix the last couple of issues related to building from a path that contains spaces.
(closes issue #11834)
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 line
This is a fix for 2 things: a problem Terry was having in OSX with null pointers, which was my fault, as I probably forgot to run the sed script last time I made mods. So, I moved the fix into the flex input itself. Then, I found when I used flex 2.5.33, that it was using __STDC_VERSION__, and that's not real good; so I added back in a DIFFERENT sed script to fix that little mess. Tested everything, a couple different ways. Hope I did no harm, at the least.
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revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
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not work for everyone, but it did for some. This set of changes makes trunk
start again for those having problems. Instead of building libresample as a
static library, it just links the object files in directly with the asterisk
binary.
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including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone)
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res_resample, and mark codec_resample as dependent upon res_resample. This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places. (I have another module
in a branch that needs it, too.)
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This commit imports libresample for use in Asterisk. It also adds a new codec
module, codec_resample. This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.
It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz
signed linear. But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.
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r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
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(not complete yet - still need to sort out dependecies on res_*)
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committed to trunk. Forgive me in advance any disruption this may cause, and please, report any problems via the bugtracker. The upside is that this can speed up large dialplans by 20 times (or more). Context, extension, and priority matching are all fairly constant-time searches. I introduce here my hashtables (hashtabs), and a regression for them. I would have used the ast_obj2 tables, but mine are resizeable, and don't need the object destruction capability. The hashtab stuff is well tested and stable. I introduce a data structure, a trie, for extension pattern matching, in which knowledge of all patterns is accumulated, and all matches can be found via a single traversal of the tree. This is per-context. The trie is formed on the first lookup attempt, and stored in the context for future lookups. Destruction routines are in place for hashtabs and the pattern match trie. You can see the contents of the pattern match trie by using the 'dialplan show' cli command when 'core set debug' has been done to put it in debug mode. The pattern tree traversal only traverses those parts of the tree that are interesting. It uses a scoreboard sort of approach to find the best match. The speed of the traversal is more a function of the length of the pattern than the number of patterns in the tree. The tree also contains the CID matching patterns. See the source code comments for details on how everything works. I believe the approach general enough that any issues that might come up involving fine points in the pattern matching algorithm, can be solved by just tweaking things. We shall see. The current pattern matcher is fairly involved, and replicating every nuance of it is difficult. If you find and report problems, I will try to resolve than as quickly as I can. The trie and hashtabs are added to the existing context and exten structs, and none of the old machinery has been removed for the sake of the multitude of functions that use them. In the future, we can (maybe) weed out the linked lists and save some space.
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namely main/Makefile .
I am unclear where decisions on the build environment (CFLAGS,
LDFLAGS, LIBS and so on) should be made - right now they are
split here and there.
As a first step in cleaning up this situation, i am trying to at
least collect all instances of each variable in one place.
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r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1 line
this update to Makefile corrects how ast_expr2f.c should be generated
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r81342 | russell | 2007-08-29 10:57:29 -0500 (Wed, 29 Aug 2007) | 3 lines
If chan_h323 is not being built, don't use g++ to do the final link of Asterisk.
(in response to a question on the asterisk-dev list)
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r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) | 34 lines
Merge changes from team/russell/iax_refcount.
This set of changes fixes problems with the handling of iax2_user and iax2_peer
objects. It was very possible for a thread to still hold a reference to one of
these objects while a reload operation tries to delete them. The fix here is to
ensure that all references to these objects are tracked so that they can't go away
while still in use.
To accomplish this, I used the astobj2 reference counted object model. This
code has been in one of Luigi Rizzo's branches for a long time and was primarily
developed by one of his students, Marta Carbone. I wanted to go ahead and bring
this in to 1.4 because there are other problems similar to the ones fixed by these
changes, so we might as well go ahead and use the new astobj if we're going to go
through all of the work necessary to fix the problems.
As a nice side benefit of these changes, peer and user handling got more efficient.
Using astobj2 lets us not hold the container lock for peers or users nearly as long
while iterating. Also, by changing a define at the top of chan_iax2.c, the objects
will be distributed in a hash table, drastically increasing lookup speed in these
containers, which will have a very big impact on systems that have a large number of
users or peers.
The use of the hash table will be made the default in trunk. It is not the default
in 1.4 because it changes the behavior slightly. Previously, since peers and users
were stored in memory in the same order they were specified in the configuration file,
you could influence peer and user matching order based on the order they are specified
in the configuration. The hash table does not guarantee any order in the container,
so this behavior will be going away. It just means that you have to be a little
more careful ensuring that peers and users are matched explicitly and not forcing
chan_iax2 to have to guess which user is the right one based on secret, host, and
access list settings, instead of simply using the username.
If you have any questions, feel free to ask on the asterisk-dev list.
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listen and manipulate the audio going through a channel.
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r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1 line
support for floating point numbers added to ast_expr2 $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp numbers. The MATH function returns fp numbers by default, so this fix is considered necessary.
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check dependencies for libraries.
Because these targets (e.g. minimime/libmmime.a) are real ones,
declaring them .PHONY would cause them to be rebuilt every time
(see e.g. SVN 64355).
As a workaround I am using the following CHECK_SUBDIR target:
CHECK_SUBDIR: # do nothing, just make sure that we recurse in the subdir/
minimime/libmmime.a: CHECK_SUBDIR
@cd minimime && $(MAKE) libmmime.a
which seems to do a better job than .PHONY (probably because
.PHONY forces the rebuild even if the recursive make does not think
it is necessary).
If this turns out to be the correct approach, we can then
merge it back into 1.4
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no longer make the symbols from res_adsi global since they don't need to be.
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without proper ASTCFLAGS.
This caused a problem with the buildinfo.o file not being able to find asterisk/build.h
This was affecting DESTDIR, but I *think* that if asterisk had never been installed before, it would've failed also.
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines
To be able to achieve the things that we would like to achieve with the
Asterisk GUI project, we need a fully functional HTTP interface with access
to the Asterisk manager interface. One of the things that was intended to be
a part of this system, but was never actually implemented, was the ability for
the GUI to be able to upload files to Asterisk. So, this commit adds this in
the most minimally invasive way that we could come up with.
A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to check
permissions of active manager sessions was added by Dwayne Hubbard. Then,
hacking this all together and do doing the modifications necessary to the HTTP
interface was done by me.
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r59573 | qwell | 2007-03-30 17:50:31 -0500 (Fri, 30 Mar 2007) | 2 lines
Add linux-uclibc host arch..."thingy". Sorry, I don't know what it's called...
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r53464 | russell | 2007-02-07 14:07:39 -0600 (Wed, 07 Feb 2007) | 4 lines
The clean target actually needs to run "distclean" on editline. This is
because we need to make sure that its configure script gets executed again,
because the CFLAGS we want to pass to editline may have changed.
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r52695 | russell | 2007-01-29 16:58:09 -0600 (Mon, 29 Jan 2007) | 2 lines
Add a missing quotation mark. This was pointed out by jcmoore on #asterisk-dev.
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r52647 | russell | 2007-01-29 15:36:56 -0600 (Mon, 29 Jan 2007) | 3 lines
ASTLDFLAGS needs to be passed to the editline configure script as LDFLAGS.
(issue #8928, zandbelt)
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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)
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r51262 | russell | 2007-01-18 15:54:23 -0600 (Thu, 18 Jan 2007) | 5 lines
Ensure that the locations given to the Asterisk configure script for ncurses,
curses, termcap, or tinfo are further passed along to the editline configure
script. This fixes some cross-compilation environments.
(issue #8637, reported by ovi, patch by me)
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r50867 | kpfleming | 2007-01-15 09:03:06 -0600 (Mon, 15 Jan 2007) | 2 lines
use the ACX_PTHREAD macro from the Autoconf macro archive for setting up compiler pthreads support... should improve portability to platforms with unusual pthreads requirements
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r49553 | kpfleming | 2007-01-04 16:51:01 -0600 (Thu, 04 Jan 2007) | 2 lines
add support for tracking thread-local-storage objects that exist via 'threadstorage' CLI commands
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r48554 | kpfleming | 2006-12-18 05:59:24 -0600 (Mon, 18 Dec 2006) | 3 lines
remove some now-unnecessary explicit includes of autoconfig.h
clean up per-file dependencies during 'make clean'
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r48525 | kpfleming | 2006-12-16 15:14:34 -0600 (Sat, 16 Dec 2006) | 2 lines
simplify dependency tracking system, using the compiler's built-in method for generating them, and only doing dependency tracking if developer mode is enabled via the configure script
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r46780 | file | 2006-11-01 13:39:47 -0500 (Wed, 01 Nov 2006) | 2 lines
Force poll() emulation for Darwin to always be on. It's too broken to consider being used. This resolves the console issue OSX users have been seeing. I would have liked to autoconf this but I haven't been able to come up with a test case that works. Que sera.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46782 f38db490-d61c-443f-a65b-d21fe96a405b
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and support linux as well (using fopencookie(), which should
be available in glibc).
Update configure.ac to check for funopen (BSD) and fopencookie(glibc),
and while we are at it also for gethostbyname_r
(the generated files need to be updated, or you need
to run bootstrap.sh yourself).
Document the new options in http.conf.sample
(names are only tentative, better ones are welcome).
At this point we can safely enable the option.
Anyone willing to try this on Sun and Apple platforms ?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45892 f38db490-d61c-443f-a65b-d21fe96a405b
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not stop the build on a fresh machine that has never had Asterisk installed on it before...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45847 f38db490-d61c-443f-a65b-d21fe96a405b
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- with AST_DEVMODE, building codecs/lpc10 fails because of lots
of warnings, and the configure step in editline fails as well.
Fix this by removing the -Werror in these steps.
- on FreeBSD (but probably on other platforms as well), the final
link of asterisk fails because AST_LIBS was not exported to the
subdirs Makefiles. Add a proper fix in the top-level Makefile
(a possible alternative way is to add "export AST_LIBS" near
the beginning of the file).
With this fix, i believe that some of the platform-specific
conditionals in main/Makefile are redundant (because they should
be already dealt with in the top level Makefile) but i don't
have a platform to check.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44081 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43996 | kpfleming | 2006-09-29 11:47:05 -0500 (Fri, 29 Sep 2006) | 2 lines
another cross-compile fix
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r43997 | kpfleming | 2006-09-29 11:52:27 -0500 (Fri, 29 Sep 2006) | 2 lines
support --without-curl in configure script
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r44008 | kpfleming | 2006-09-29 13:25:49 -0500 (Fri, 29 Sep 2006) | 2 lines
don't abuse CFLAGS and LDFLAGS for build of Asterisk components, because they are also then used for non-Asterisk components (like menuselect); use our own variables instead
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r44011 | kpfleming | 2006-09-29 13:40:17 -0500 (Fri, 29 Sep 2006) | 2 lines
missed one conversion to ASTCFLAGS
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r44012 | kpfleming | 2006-09-29 13:49:07 -0500 (Fri, 29 Sep 2006) | 2 lines
yet another place where we were not using the correct CFLAGS by default
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44013 f38db490-d61c-443f-a65b-d21fe96a405b
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#7047 reported by maddison)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43830 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43450 | kpfleming | 2006-09-21 16:21:29 -0500 (Thu, 21 Sep 2006) | 2 lines
add another attempt to strip non-API symbols from the final binary... script will need to be extended to work on non-Linux systems
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43451 f38db490-d61c-443f-a65b-d21fe96a405b
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