https://origsvn.digium.com/svn/asterisk/branches/1.4
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r82867 | russell | 2007-09-18 15:56:43 -0500 (Tue, 18 Sep 2007) | 10 lines
Fix a memory leak that can occur on systems under higher load. The issue is
that when events are appended to the master event queue, they use the number
of active sessions as a use count so it will know when all active sessions
at the time the event happened have consumed it. However, the handling of
the number of sessions was not properly synchronized, so the use count was
not always correct, causing an event to disappear early, or get stuck in
the event queue for forever.
(closes issue #9238, reported by bweschke, patch from Ivan, modified by me)
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r82337 | russell | 2007-09-13 13:45:59 -0500 (Thu, 13 Sep 2007) | 4 lines
Only compile in tracking astobj2 statistics if dev-mode is enabled. Also, when
dev mode is enabled, register the CLI command that can be used to run the astobj2
test and print out statistics.
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r82261 | murf | 2007-09-11 14:36:15 -0600 (Tue, 11 Sep 2007) | 1 line
this change should fix issue # 10659 -- what I worry about is how many other bug reports it may generate. Hopefully, we can please the/a majority. Hopefully. We shall see. Calls not marked ANSWERED and with only one channel name will not be posted. This should eliminate the double CDR's.
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Reported by: junky
Patches:
count_showconn.diff uploaded by junky (license 177)
Provide a count of connected users to manager.
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Reported by: snuffy
Patches:
minivm.diff uploaded by snuffy (license 35)
Instead of using err (which is not available under Solaris) use fdprintf with stderr.
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Note: this is slightly different than the initial patch, because I felt
that using res <= 0 would be a change in behavior.
Closes issue #10687, patch by junky
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ao2_hash_fn and ao2_callback_fn typedefs, in preparation
to more cleanup of the _search_flags
Please do not merge this change to 1.4 yet - there are no
functional changes anyways.
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r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | 11 lines
Fix an issue that can occur when you do an attended transfer to parking. If
you complete the transfer before the announcement of the parking spot finishes,
then the channel being parked will hear the remainder of the announcement.
These changes make it so that will not happen anymore.
Basically, res_features sets a flag on the channel is playing the announcement
to so that the file streaming core knows that it needs to watch out for a
channel masquerade, and if it occurs, to abort the announcement.
(closes BE-182)
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them directly in Asterisk. For platforms that need them (like my mac), you
will get a linker error due to the functions being included twice.
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r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) | 4 lines
Remove the typedefs on ao2_container and ao2_iterator. This is simply because
we don't typedef objects anywhere else in Asterisk, so we might as well make
this follow the same convention.
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r81392 | murf | 2007-08-30 15:11:48 -0600 (Thu, 30 Aug 2007) | 1 line
via issue 10599, where 'CDR already initialized' messages are being generated. Since all channels will have an init'd CDR attached at creation time, this message is now particularly useless. Removed.
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Reported by: jmls
Patches:
pbx.diff uploaded by jmls (license 141)
Add REASON dialplan variable for when an originated call fails and the failed extension is executed.
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* Convert some spaces to tabs in func_volume
* Add a note in channel.h making it clear that none of the datastore API calls
lock the channel they are given, so the channel should be locked before
calling the functions that take a channel argument.
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Reported by: nic_bellamy
Patches:
2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by nic_bellamy (license 213)
Add support for configurable file locking methods. The default is "lockfile",
which is the old behavior. There is an additional option, "flock", which is
intended for use in situations where the lockfile method will not work, such as
with SMB/CIFS mounts.
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r80820 | russell | 2007-08-24 15:24:05 -0500 (Fri, 24 Aug 2007) | 7 lines
Improve the debouncing logic in the DTMF detector to fix some reliability
issues. Previously, this code used a shift register of hits and non-hits.
However, if the start of the digit isn't clean, it is possible for the
leading edge detector to miss the digit. These changes replace the flawed
shift register logic and also does the debouncing on the trailing edge as well.
(closes issue #10535, many thanks to softins for the patch)
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r80789 | murf | 2007-08-24 12:52:15 -0600 (Fri, 24 Aug 2007) | 1 line
From a complaint by jmls, I realize that the message in cdr_disposition is unnecessary. To get failure disposition, just return -1; no use having more than one case do that.
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r80424 | russell | 2007-08-22 17:40:27 -0500 (Wed, 22 Aug 2007) | 10 lines
When converting this code to use the list macros, I changed it so objects are
added to the head of a bucket instead of the tail. However, while looking over
code with mmichelson, we noticed that the algorithm used in ao2_iterator_next
requires that items are added to the tail. This wouldn't have caused any huge
problem, but it wasn't correct. It meant that if an object was added to a
container while you were iterating it, and it was added to the same bucket that
the current element is in, then the new object would be returned by
ao2_iterator_next, and any other objects in the bucket would be bypassed in
the traversal.
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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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r80166 | murf | 2007-08-21 10:36:34 -0600 (Tue, 21 Aug 2007) | 1 line
This patch solves problem 1 in 8126; it should not slow down the alaw codec, but should prevent signal degradation via multiple trips thru the codec. Fossil estimates the twice thru this codec will prevent fax from working. 4-6 times thru would result hearable, noticeable, voice degradation.
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(closes issue #10430)
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r79904 | qwell | 2007-08-17 14:12:19 -0500 (Fri, 17 Aug 2007) | 11 lines
Don't send a semicolon over the wire in sip notify messages.
Caused by fix for issue 9938.
I basically took the code that existed before 9938 was fixed, and
copied it into a new function - ast_unescape_semicolon
There should be very few places this will be needed (pbx_config
does NOT need this (see issue 9938 for details))
Issue 10430, patch by me, with help/ideas from murf (thanks murf).
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in place of a very common construct. I also used it in a number of places
in chan_sip.
if (id > -1)
ast_sched_del(sched, id);
id = ast_sched_add(sched, ...);
changes to:
ast_sched_replace(id, sched, ...);
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Reported by: pj
Two of the three places ast_waitfor_nandfds could branch off to did not clear outfd and exception. If the calling function did not clear these there was a chance they could get a false positive on testing to see whether they were set.
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The way a device state change propagates is kind of silly, in my opinion. A
device state provider calls a function that indicates that the state of a
device has changed. Then, another thread goes back and calls a callback for
the device state provider to find out what the new state is before it can go
send it off to whoever cares.
I have changed it so that you can include the state that the device has changed
to in the first function call from the device state provider. This removes the
need to have to call the callback, which locks up critical containers to go find
out what the state changed to.
This change set changes the "simple" device state providers to use the new method.
This includes parking, meetme, and SLA.
I have also mostly converted chan_agent in my branch, but still have some more
things to think through before presenting the plan for converting channel drivers
to ensure all of the right events get generated ...
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Reported by: klaus3000
Clean up AST_FORMAT_LIST list. It may have mattered in the old days to have undefined entries but these days it does not.
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r78103 | mmichelson | 2007-08-03 15:25:22 -0500 (Fri, 03 Aug 2007) | 7 lines
Changed the behavior of sip's realtime_peer function to match the corresponding way of matching for non-realtime peers.
Now matches are made on both the IP address and port number, or if the insecure setting is set to "port" then just match on the
IP address.
In order to accomplish this, I also added a new API call, ast_category_root, which returns the first variable of an ast_category struct
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r78172 | file | 2007-08-06 12:27:24 -0300 (Mon, 06 Aug 2007) | 4 lines
(closes issue #10355)
Reported by: wdecarne
Now that we pass through RTP timestamp information we need to make the allowed timestamp skew considerably less. There are situations where a source may change and due to the timestamp difference the receiver will experience an audio gap since we did not indicate by setting the marker bit that the source changed.
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r78095 | russell | 2007-08-03 14:39:49 -0500 (Fri, 03 Aug 2007) | 28 lines
Add some improvements to lock debugging. These changes take effect
with DEBUG_THREADS enabled and provide the following:
* This will keep track of which locks are held by which thread as well as
which lock a thread is waiting for in a thread-local data structure. A
reference to this structure is available on the stack in the dummy_start()
function, which is the common entry point for all threads. This information
can be easily retrieved using gdb if you switch to the dummy_start() stack
frame of any thread and print the contents of the lock_info variable.
* All of the thread-local structures for keeping track of this lock information
are also stored in a list so that the information can be dumped to the CLI
using the "core show locks" CLI command. This introduces a little bit of a
performance hit as it requires additional underlying locking operations
inside of every lock/unlock on an ast_mutex. However, the benefits of
having this information available at the CLI is huge, especially considering
this is only done in DEBUG_THREADS mode. It means that in most cases where
we debug deadlocks, we no longer have to request access to the machine to
analyze the contents of ast_mutex_t structures. We can now just ask them
to get the output of "core show locks", which gives us all of the information
we needed in most cases.
I also had to make some additional changes to astmm.c to make this work when
both MALLOC_DEBUG and DEBUG_THREADS are enabled. I disabled tracking of one
of the locks in astmm.c because it gets used inside the replacement memory
allocation routines, and the lock tracking code allocates memory. This caused
infinite recursion.
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r77945 | murf | 2007-08-02 12:21:40 -0600 (Thu, 02 Aug 2007) | 9 lines
Merged revisions 77942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r77942 | murf | 2007-08-02 11:56:37 -0600 (Thu, 02 Aug 2007) | 1 line
This patch hopefully solves 10141; The user is running with it, and it doesn't appear to harm asterisk's operation, and may prevent a crash. I'll store it in 1.2, as we have shut down support on 1.2, but since I developed the patch before support finished, and it might affect 1.4 and trunk, I'm going ahead with it.
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(closes issue #10083)
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r77795 | qwell | 2007-07-30 15:17:08 -0500 (Mon, 30 Jul 2007) | 6 lines
Applications like SayAlpha() should not hang up the channel if you
request an "unknown" character such as a comma.
Instead, skip the character and move on.
Issue 10083, initial patch by jsmith, modified by me.
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inside of a function. (Yes, it would!) Replace it with a note that explains
why it can't be done using the way that the AST_THREADSTORAGE macro is
currently defined.
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r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) | 16 lines
(closes issue #10301)
Reported by: fnordian
Patches:
asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
Additional changes by me
Fix some problems in channel_find_locked() which can cause an infinite loop.
The reference to the previous channel is set to NULL in some cases. These changes
ensure that the reference to the previous channel gets restored before needing
it again.
I'm not convinced that the code that is setting it to NULL is really the right
thing to do. However, I am making these changes to fix the obvious problem
and just leaving an XXX comment that it needs a better explanation that what
is there now.
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r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6 lines
(closes issue #10301)
Reported by: fnordian
Patches:
asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
Restore previous behavior where if we failed to lock the channel we wanted we would return to exactly the same point as if we had just reentered the function.
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r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4 lines
(closes issue #10302)
Reported by: litnialex
If a DTMF end frame comes from a channel without a begin and it is going to a technology that only accepts end frames (aka INFO) then use the minimum DTMF duration if one is not in the frame already.
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r77380 | mmichelson | 2007-07-26 15:35:17 -0500 (Thu, 26 Jul 2007) | 7 lines
Fixes to get ast_backtrace working properly. The AST_DEVMODE macro was never defined so the majority of ast_backtrace never
attempted compilation. The makefile now defines AST_DEVMODE if configure was run with --enable-dev-mode. Also, changes were
made to acccomodate 64 bit systems in ast_backtrace.
Thanks to qwell, kpfleming, and Corydon76 for their roles in allowing me to get this committed
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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).
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Reported by: snuffy
Patches:
doxygen-updates.diff uploaded by snuffy (license 35)
Another big batch of doxygen documentation updates
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Reported by: bbryant
Patches:
20070720__core_debug_by_file.patch uploaded by bbryant (license 36)
(with some modifications by me)
Tested by: russell, bbryant
This set of changes introduces the ability to set the core debug or verbose
levels on a per-file basis. Interestingly enough, in 1.4, you have the ability
to set core debug for a single file, but that functionality was accidentally
lost in the conversion of the CLI commands to the new format.
This patch improves upon what was in 1.4 by letting you set it for more than 1
file, and by also supporting verbose.
*** Janitor Project ***
This patch also introduces a new macro, ast_verb(), which is similar
to ast_debug(). Setting the per file verbose value only works for messages that
use this macro. Converting existing uses of ast_verbose() can be done like:
if (option_debug > 2)
ast_verbose(VERBOSE_PREFIX_3 "Something useful\n");
...
ast_verb(3, "Something useful\n");
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r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) | 6 lines
Use the define that specifies the default length of an artificially created
DTMF digit in the ast_senddigit() function. The define is set to 100ms by
default, which is the same thing that this function was using. But, using
the define lets changes take effect in this case, as well as the others where
it was already used.
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(e.g. chan_sip.c in a subsequent commit).
Obviously exposing the internals of a data structure is far from ideal
(especially in a case like this where the implementation is very
inefficient and will need to be changed at some point).
On the other hand, it was also unclear what additional APIs should
we provide instead, and because exposing the stucture has no impact
on source and binary compatibility, this seemed to me the best option at
this time.
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r75403 | russell | 2007-07-17 15:01:12 -0500 (Tue, 17 Jul 2007) | 12 lines
(closes issue #10209)
Reported by: juggie
Patches:
10209-trunk-2.patch uploaded by juggie
Tested by: juggie, blitzrage
In ast_pbx_run(), mark a channel as hung up after an application returned -1,
or when it runs out of extensions to execute. This is so that code can detect
that this channel has been hung up for things like making sure DeadAGI is used
on actual dead channels, and is beneficial for other things, like making sure
someone doesn't try to start spying on a channel that is about to go away.
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in a consistent way. This is meant to replace the custom code
which is repeated all over the place in the various files when
parsing config files, CLI entries and other string information.
Right now the code supports parsing int32, uint32 and sockaddr_in with
optional default values and bound checks. It contains minimal error
checking, but that can be easily extended as the need arises.
Being a new API i am introducing this only in trunk, though I believe
that once the interface has been ironed out it might become a
worthwhile addition to 1.4 as well - basically, the first time
we will need to fix a piece of argument parsing code, we might as
well bring in this change and use the new API instead.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
sockets other than RTP ones.
The main change is a new API function in main/rtp.c (see there
for a description)
int ast_stun_request(int s, struct sockaddr_in *dst,
const char *username, struct sockaddr_in *answer)
which can be used to send an STUN request on a socket, and
optionally wait for a reply and store the STUN_MAPPED_ADDRESS
into the 'answer' argument (obviously, the version that
waits for a reply is blocking, but this is no different
from DNS resolutions).
Internally there are minor modifications to let stun_handle_packet()
be somewhat configurable on how to parse the body of responses.
At the moment i am not committing any change to the clients,
but adding STUN client support is extremely simple, e.g. chan_sip.c
could do something like this:
+ add a variable to store the stun server address;
static struct sockaddr_in stunaddr = { 0, }; /*!< stun server address */
+ add code to parse a config file of the form "stunaddr=my.stun.server.org:3478"
(not shown for brevity);
+ right after binding the main sip socket, talk to the stun server to
determine the externally visible address
if (stunaddr.sin_addr.s_addr != 0)
ast_stun_request(sipsock, &stunaddr, NULL, &externip);
so now 'externip' is set with the externally visible address.
so it is really trivial.
Similarly ast_stun_request could be called when creating the RTP
socket (possibly adding a struct sockaddr_in field in the struct
ast_rtp to store the externalip).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
+ mark a potentially dangerous write-past-end-of-buffer
+ localize some variables in the block generating stun replies.
As before, not ready yet for a merge to 1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_rtp_new_with_bindaddr():
1. add comments to the logic of the main loop;
2. use a common exit point on failure so the cleanup is done only in one place;
3. handle failures in rtp_socket() in the main loop of the function;
No functional changes except for #3 above, so it is not yet
worthwhile merging this and other changes to 1.4
Once the cleanup work on this file will be complete (which among
other things should include some extensions to the stun support)
it might be a good thing to push all the changes to 1.4
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(which is very little, at the moment).
Eventually, when the functionality is extended, the changes can be merged
back to 1.4. At the moment this is pointless.
Note, this change is whitespace only.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10133)
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r74374 | qwell | 2007-07-10 13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines
Merged revisions 74373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 lines
Use res_ndestroy on systems that have it. Otherwise, use res_nclose.
This prevents a memleak on NetBSD - and possibly others.
Issue 10133, patch by me, reported and tested by scw
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for extracting application, function, manager, and agi documentation is the wrong
one to take. The most severe problem is that the output depends on which modules
are loaded as well as compile time options, which both determine which parts are
available.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
*must* write the file to the FILE *, and not the raw fd. Otherwise, it breaks
TLS support.
Thanks to rizzo for catching this!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Use weaker error checking because we have some automatically generated
files. However just mask out -Werror, because other warnings below:
-Wundef -Wstrict-prototypes -Wmissing-declarations
-Wmissing-prototypes
may actually be important and spot out real bugs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Handle transferring large files from the built-in http server. Previously, the
code attempted to malloc a block as large as the file itself. Now it uses the
sendfile() system call so that the file isn't copied into userspace at all if
it is available. Otherwise, it just uses a read/write of small chunks at a time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r72383 | bbryant | 2007-06-27 18:29:14 -0500 (Wed, 27 Jun 2007) | 11 lines
Merged revisions 72373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r72373 | bbryant | 2007-06-27 18:22:13 -0500 (Wed, 27 Jun 2007) | 3 lines
Reinstating patch. This actually fixes the problem, however I was running a development branch without it and mistakenly thought it wasn't fixed.
Fixes issue #10010, and #9654: 100% CPU usage caused by an asterisk console losing it's controlling terminal.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r72257 | file | 2007-06-27 16:25:24 -0400 (Wed, 27 Jun 2007) | 10 lines
Merged revisions 72256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 lines
I may possibly get shot for doing this... but... defer CDR processing until after the channel has been dealt with. This should eliminate all of the issues with channels going funky (SIP/PRI) when you are posting CDRs to a database that is either slow or unavailable and do not want to enable batching.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun 2007) | 1 line
My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone.
This is for bug 10016. (plus a small fix to rtp, to elim a compiler warning (dev mode))
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r71066 | bbryant | 2007-06-22 09:53:08 -0500 (Fri, 22 Jun 2007) | 18 lines
Merged revisions 71064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r71064 | bbryant | 2007-06-22 09:39:34 -0500 (Fri, 22 Jun 2007) | 10 lines
Fixed infinite loop when controlling terminal was lost
and return value of input function wasn't checked for
errors. This would cause 100% cpu to be taken up.
(closes issue #9654, issue #10010)
Reported by: mnicholson, and eserra
Idea for the patch from mnicholson, patched by me
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
This permission was discussed on the -dev mailing list some months back.
Issue 8613, patch by johann8384, with some minor changes by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Jason Parker
Russell Bryant
Doug Bailey
Joshua Colp
Steve Murphy
Tilghman Lesher
Mark Michelson
Luigi Rizzo
Kevin P. Fleming
Brett Bryant