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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92422 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91114 f38db490-d61c-443f-a65b-d21fe96a405b
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90873 f38db490-d61c-443f-a65b-d21fe96a405b
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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- Fix typo in chan_sip
- Remove changes to caller ID structure, moving it to branch (russellb)
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89531 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@89522 f38db490-d61c-443f-a65b-d21fe96a405b
tilghman