equivalent to the check done by ast_verb, I wrote a macro, VERBOSITY_LEVEL, which does this
check. I did a quick look in the source and used this macro in some places where option_verbose
was used.
I also converted some verbose messages in logger.c to use ast_verb instead of ast_verbose.
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r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13 Dec 2007) | 7 lines
When compiling with DETECT_DEADLOCKS, don't spam the CLI with messages
about possible deadlocks. Instead just print the intended single message every
five seconds.
(closes issue 11537, reported and patched by dimas)
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structures missing. Patched configure to check for this stuff and
put a #ifdef around the offending code in chan_zap. Thanks to file
for overseeing this.
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get_unaligned functions as const
* In event.c, use get_unaligned_uint32() in a couple of places to fix issues on
architectures that don't allow unaligned access
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r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07 Dec 2007) | 6 lines
Fix another bug in the DEBUG_THREADS code. The ast_mutex_init() function had
the mutex attribute object marked as static. This means that multiple threads
initializing locks at the same time could step on each other and end up with
improperly initialized locks.
(found when tracking down locking issues related to issue #11080)
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r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07 Dec 2007) | 6 lines
I love fixing lock related errors in the lock debugging code. That's about as
ironic as it gets in Asterisk programming land. Anyway, I spotted this bug while
trying to track down why systems are locking up and acting weird in issue #11080.
The mutex attribute object was marked as static in this function when it should
not have been.
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r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04 Dec 2007) | 11 lines
Fix some crashes in chan_iax2 that were reported as happening on Mac systems.
It turns out that the problem was the Mac version of the ast_atomic_fetchadd_int()
function. The Mac atomic add function returns the _new_ value, while this function
is supposed to return the old value. So, the crashes happened on unreferencing
objects. If the reference count was decreased to 1, ao2_ref() thought that it
had been decreased to zero, and called the destructor. However, there was still
an outstanding reference around.
(closes issue #11176)
(closes issue #11289)
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This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify.
The mwimonitor option enables MWI monitoring. When the MWI state on a line changes,
then the script specified by mwimonitornotify will be executed for custom handling
of the state change, similar to the externnotify option of voicemail.conf.
Also, when the MWI state on an FXO line changes, an internal Asterisk event is
generated to indicate the new state of the associated mailbox. That may, any
module that cares about MWI information will get notified and can handle it
just as if app_voicemail had sent this notification.
(BE-253, original patch from markster, with some minor modifications by me to
add comments, documentation, and internal event support)
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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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This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
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r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines
Change the behavior of ao2_link(). Previously, in inherited a reference.
Now, it automatically increases the reference count to reflect the reference
that is now held by the container.
This was done to be more consistent with ao2_unlink(), which automatically
releases the reference held by the container. It also makes it so it is
no longer possible for a pointer to be invalid after ao2_link() returns.
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r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines
This set of changes is to make some callerID handling thread-safe.
The ast_set_callerid() function needed to lock the channel. Also, the handlers
for the CALLERID() dialplan function needed to lock the channel when reading
or writing callerid values directly on the channel structure.
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This was mostly to note whether a channel needed to be locked or not before
calling these functions. However, I added some other things, too.
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r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | 4 lines
- update documentation for some of the goto functions to note that they
handle locking the channel as needed
- update ast_explicit_goto() to lock the channel as needed
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Also by accident fixed a bad typo by a previous committer, which actually made video calls
not work fully...
Merged revisions 89630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines
If we get a codec offer using a well-known payload type, but using it for another
codec that we don't know, Asterisk did not remove that codec from the list.
With this patch, we remove the codec from audio and video rtp objects and
deny it ever existed. Thanks to lasse for testing.
(closes issue #11376)
Reported by: lasse
Patches:
bug11376.txt uploaded by oej (license 306)
Tested by: lasse
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except with an additional boolean arg.
A hack such as:
foo ? S_OR(bar, "baz") : "baz"
becomes:
S_COR(foo, bar, "baz")
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r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line
closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages.
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r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines
We previously attempted to use the ESCAPE clause to set the escape delimiter to
a backslash. Unfortunately, this does not universally work on all databases,
since on databases which natively use the backslash as a delimiter, the
backslash itself needs to be delimited, but on other databases that have no
delimiter, backslashing the backslash causes an error.
So the only solution that I can come up with is to create an option in res_odbc
that explicitly specifies whether or not backslash is a native delimiter. If
it is, we use it natively; if not, we use the ESCAPE clause to make it one.
Reported by: elguero
Patch by: tilghman
(Closes issue #11364)
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so we can handle multiple formats properly. This is not carved in stone,
but a proposal to start with.
We need to add support for transliterations as well as UTF8 handling,
propably with libiconv. Murf is looking into that for the dialplan.
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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.
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to compat.h so it is always available - hopefully this will let
us reduce the number of inclusions of channel.h and frame.h
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include/asterisk/autoconfig.h. Also, move the conditional include of sys/poll.h
or asterisk/poll-compat.h into asterisk/config.h instead of the two headers it
existed in before.
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and the like (this could be used for gtk and gtk2 as well)
Other files: add tests for sdl, sdl_image and avcodec and regenerate
configure and autoconfig.h.in
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comments on the content of these two files.
utils.h (which is included in over 150 files) contains a lot of
unrelated functions which require the inclusion of a large number
of other headers. At some point we should partition its content
in a better way.
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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.).
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r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007) | 4 lines
To help combat problems where people build external modules (asterisk-addons or others) and then change the build options of the Asterisk build in a way that makes the incompatible without warning, this commit introduces an MD5 signature of the important build-time options and includes that signature into modules when they are built. When the loader loads one of these modules and notices the problem, it will emit a warning to console and refuse to initialize the module, as doing so could cause the system to be unstable or even crash.
If you upgrade to this version of Asterisk, you must rebuild *all* of your modules that came from other sources before trying to run this version. If you are using Digium's G.729 binary codec module, you will need v33 or newer.
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through ast_mutex primitives.
To detect all occurrences, I have renamed the lock field in struct ast_channel
so it is clear that it shouldn't be used directly.
There are some uses in res/res_features.c (see details of the diff)
that are error prone as they try and lock two channels without
caring about the order (or without explaining why it is safe).
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This prevents modifying the strings in the stored variables,
and catched a few instances where this was actually done.
Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are
chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049
I may have missed some instances for modules that do not build here.
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- handle memory allocation failures
- add an ast_ prefix to a publicly exported function
- put curly braces in the right places
- add a bunch of spaces where they should be be used
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Also fix a common typo I kept seeing (arguement) in various files.
Closes issue #11222, patch by snuffy (with arguement > argument by me).
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
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- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
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r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines
After seeing crashes related to channel variables, I went looking around at the
ways that channel variables are handled. In general, they were not handled in
a thread-safe way. The channel _must_ be locked when reading or writing from/to
the channel variable list.
What I have done to improve this situation is to make pbx_builtin_setvar_helper()
and friends lock the channel when doing their thing. Asterisk API calls almost
all lock the channel for you as necessary, but this family of functions did not.
(closes issue #10923, reported by atis)
(closes issue #11159, reported by 850t)
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r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06 Nov 2007) | 8 lines
Remove some checks to see if locks are initialized from the non-DEBUG_THREADS
versions of the lock routines. These are incorrect for a number of reasons:
- It breaks the build on mac.
- If there is a problem with locks not getting initialized, then the proper
fix is to find that place and fix the code so that it does get initialized.
- If additional debug code is needed to help find the problem areas, then this
type of things should _only_ be put in the DEBUG_THREADS wrappers.
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r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) | 7 lines
Merge changes from asterisk/team/kpfleming/SRV-priority-handling
Previously, the SRV record support in Asterisk was broken. There was no
guarantee on what record Asterisk would choose to actually use. This set of
changes improves the situation by ensuring that Asterisk will choose the
highest priority record.
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details and examples are in include/asterisk/stringfields.h.
Not applicable to older branches except for 1.4 which will
receive a fix for the routines that free memory pools.
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r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1 line
closes issue #11086 where a user complains that references to following contexts report a problem; The problem was REALLy that he was referring to empty contexts, which were being ignored. Reporter stated that empty contexts should be OK. I checked it out against extensions.conf, and sure enough, empty contexts ARE ok. So, I removed the restriction from AEL. This, though, highlighted a problem with multiple contexts of the same name. This should be OK, also. So, I added the extend keyword to AEL, and it can preceed the 'context' keyword (mixed with 'abstract', if nec.). This will turn off the warnings in AEL if the same context name is used 2 or more times. Also, I now call ast_context_find_or_create for contexts now, instead of just ast_context_create; I did this because pbx_config does this. The 'extend' keyword thus becomes a statement of intent. AEL can now duplicate the behavior of pbx_config,
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r86836 | russell | 2007-10-22 16:36:12 -0500 (Mon, 22 Oct 2007) | 9 lines
If lock tracking is not enabled, then we can not attempt to log any mutex
failures. If so, we could end up in infinite recursion. The only lock that
is affected by this is a mutex in astmm.c used when MALLOC_DEBUG is enabled.
(closes issue #11044)
Reported by: ys
Patches:
lock.h.diff uploaded by ys (license 281)
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r85532 | russell | 2007-10-13 00:24:33 -0500 (Sat, 13 Oct 2007) | 8 lines
Properly handle the case where read() may return the text for more than one
CLI command at once for a remote console.
(closes issue #10888)
Reported by: jamesgolovich
Patches:
asterisk-climultiple.diff.txt uploaded by jamesgolovich (license 176)
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r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) | 10 lines
The channel needs to stay locked while running timer callbacks, as they access
and modify channel data that may change elsewhere. I went through every timer
callback in the source tree to make sure that none of them did any additional
locking that could introduce deadlocks, and all is well.
(closes issue #10765)
Reported by: Ivan
Patches:
ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license 229)
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r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16 Oct 2007) | 16 lines
Some locking errors exposed the fact that the lock debugging code itself was
not thread safe. How ironic! Anyway, these changes ensure that the code that
is accessing the lock debugging data is thread-safe.
Many thanks to Ivan for finding and fixing the core issue here, and also
thanks to those that tested the patch and provided test results.
(closes issue #10571)
(closes issue #10886)
(closes issue #10875)
(might close some others, as well ...)
Patches: (from issue #10571)
ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license 229)
- a few small changes by me
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r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) | 12 lines
Fix an issue with console verbosity when running asterisk -rx to execute a command
and retrieve its output. The issue was that there was no way for the main Asterisk
process to know that the remote console was connecting in the -rx mode. The way that
James has fixed this is to have all remote consoles muted by default. Then, regular
remote consoles automatically execute a CLI command to unmute themselves when they
first start up.
(closes issue #10847)
Reported by: atis
Patches:
asterisk-consolemute.diff.txt uploaded by jamesgolovich (license 176)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10 Oct 2007) | 6 lines
I introduced a new member to the ast_filestream struct in 1.4.12, but put it
in the middle of the struct, instead of at the end. One of the Debian folks,
paravoid, pointed out that this breaks binary compatability with modules
compiled against older headers. So, I'm moving the new member to the end
of the struct to resolve the situation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
No real Jingle implementation being available, testing was made using
two Asterisk servers relaying SIP calls over their Jingle channels:
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Thus, it was possible to test the code in both ways, and make the
Jingle channel comply with the latest specifications. No sound available yet.
Main modifications include :
- modified the 'jingle_candidate' structure and the
'jingle_create_candidates' function according to XEP-0176 ;
- modified the 'jingle_action' function in order to properly terminate
a Jingle session, in conformance with XEP-0166 ;
- modified username format used in STUN requests ;
- actually make the bindaddr configuration field useable.
Todo :
- set audio paths up (no native bridging) ;
- make the CLI gtalk functions available to jingle ;
- clean up the storage space used in strings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines
gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) | 11 lines
Add a new patch to handle interrupting the fgets() call when using FastAGI.
This version of the patch maintains the original behavior of the code when
not using FastAGI.
(closes issue #10553)
Reported by: juggie
Patches:
res_agi_fgets-4.patch uploaded by juggie (license 24)
res_agi_fgets_1.4svn.patch uploaded by juggie (license 24)
Slight mods by me
Tested by: juggie, festr
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Mark Michelson
Joshua Colp
Tilghman Lesher
Olle Johansson
Russell Bryant
Luigi Rizzo
Jason Parker
Steve Murphy
Kevin P. Fleming
Philippe Sultan
Dwayne M. Hubbard