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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
what build options were used. We agreed that we should remove this before
making a 1.4 release, and then we can put it back in. Then, we can take a
month or so to play around with it to get it how we want it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@89325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and ast_string_field_free_all to ast_string_field_reset_all
to avoid misuse (due to too similar names and an error in
documentation). Fix two related memory leaks in app_meetme.
No need to merge to trunk, different fix already applied there.
Not applicable to 1.2
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@88471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@86836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@86330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CLI command at once for a remote console.
(closes issue #10888)
Reported by: jamesgolovich
Patches:
asterisk-climultiple.diff.txt uploaded by jamesgolovich (license 176)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@85316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@82929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@81599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
after we already looked it up by name. This causes broken behavior if there is
more than one feature defined with the same digit pattern.
(closes issue #10539, reported by bungalow, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@80573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the peers and users are being stored in a linked list, that they go in the
list in the same order that the older code used. This is necessary to maintain
the behavior of which peers and users get matched when traversing the container.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@80497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
implies that is possible to miss an object or see an object twice while
iterating. After looking through the code and talking with mmichelson, I have
documented the exact conditions under which this can happen (which are rare and
harmless in most cases).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@80426 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@80362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@79904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
track mutexes unless they were declared using AST_MUTEX_DEFINE_STATIC. Locks
initialized with ast_mutex_init() were not tracked. It should work now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@78095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
still can't build *_odbc.so!", check for ltdl directly, instead of just listing
it as another library to include in the unixodbc check in the configure script.
This also makes ltdl show up as a dependency in menuselect so people know what
to go install. (related to issue #9989, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to a bug in the iksemel library, this will not work if you are using GTLS
in the connection. That's being investigated. If you figure out a way to handle
that without us having to patch iksemel, let us know in the bug report. Thanks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines
We have some bug reports showing crashes due to a double free of a channel.
Add a sanity check to ast_channel_free() to make sure we don't go on trying
to free a channel that wasn't found in the channel list.
(issue #8850, and others...)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67716 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Every time I tried to go debug it by adding some debug output, the behavior
would change. It turns out I wasn't crazy. I had the following piece of code:
if (remove)
AST_LIST_REMOVE_CURRENT(...);
Well, AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my conditional
statement didn't do much good at all. It always ran at least all of the
macro minus the first statement, so I was seeing list entries magically
disappear when they weren't supposed to.
After many hours of debugging, I have come to this extremely irritating fix. :)
(issues #9581, #9497)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
all of the modules. "stop now" is considered a non-graceful shutdown and will
not go through this process.
(issue #9804, reported by chrisost, patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the asterisk-dev mailing list. I changed the enforced minimum length of a
digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in
between digits. These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@61781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines
Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list).
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in'
(the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Originally, I put in the documentation that only Zap interfaces would be
supported on the trunk side. However, after a discussion with Qwell, we came
up with a way to make IP trunks work as well, using some things already in
Asterisk. So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
in SLA. The station's channel needs to be passed to the dial API when
dialing the trunk.
* Change a WARNING message to DEBUG in channel.h. This message is of no use
to users.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Russell Bryant
Olle Johansson
Steve Murphy
Tilghman Lesher
Kevin P. Fleming
Luigi Rizzo
Dwayne M. Hubbard
Jason Parker
Philippe Sultan
Mark Michelson
Joshua Colp
Nadi Sarrar