The WebSocket code would allocate, on the stack, a string large enough
to hold a key provided by the client, and the WEBSOCKET_GUID. If the key
is NULL, this causes a segfault. If the key is too large, it could
overflow the stack.
This patch checks the key for NULL and checks the length of the key to
avoid stack smashing nastiness.
(closes issue ASTERISK-21825)
Reported by: Alfred Farrugia
Tested by: Alfred Farrugia, David M. Lee
Patches:
issueA21825_check_if_key_is_sent.patch uploaded by Walter Doekes (license 5674)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes three memory leaks
* When we load a module with the LOAD_PRIORITY flag, we remove its entry from
the load order list. Unfortunately, we don't free the memory associated with
entry in the list. This patch corrects that and properly frees the memory
for the module in the list.
* When adding a custom format (such as SILK or CELT), the routine for adding
the format was leaking a reference. RAII_VAR cleans this up properly.
* We now de-ref the channel_snapshot appropriately when an endpoint is
disposed of
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Merged revisions 391507 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes two memory leaks:
* A memory leak in packing channels into a multi-channel blob payload when
publishing dial messages. The multi-channel blob payload does not steal
the references - this approach was chosen because it works well with the
RAII_VAR macro. Unfortunately, this does mean that you actually have to use
the RAII_VAR macro (or manually deref it yourself)
* RTP instances returned as a result of one of the glue operations are ref
counted and have to be de-ref'd appropriately. We now do that, as saying
that we should do it and then not would be silly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When checking compatability for the native RTP bridge technology there is a
race condition between clearing framehooks that are destroyed when leaving
certain bridges with certain technologies (such as bridge_native_rtp) and
joining bridges with the bridge_native_rtp technology. Yes, that means a
channel in a native RTP bridge could move to another native RTP bridge and
be considered incompatible with the new native RTP bridge causing it to
revert to a simple bridge technology0. This fixes that bug by ignoring
framehooks that have been marked for destruction when checking for
compatibility with the bridge_native_rtp technology.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
native_rtp_bridge_get can return any result from the ast_rtp_glue_result
enumerator and the join/leave functions for bridge_native_rtp seem to assume
that if the result wasn't local that it was remote. Meanwhile forbid can be
returned by that function which can mean certain glue pointers are NULL. Then
when the join/leave functions try to use members of that pointer, boom.
Segfault.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a Stasis message type is defined in a loadable module, handling
those messages for AMI and res_stasis events can be cumbersome.
This patch adds a vtable to stasis_message_type, with to_ami and
to_json virtual functions. These allow messages to be handled
abstractly without putting module-specific code in core.
As an example, the VarSet AMI event was refactored to use the to_ami
virtual function.
(closes issue ASTERISK-21817)
Review: https://reviewboard.asterisk.org/r/2579/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
JSON objects are reference stealing. Hence, if you've RAII_VAR'd some
subobject and want to pack it into another JSON object, you have to bump
the reference count. Using the 'O' option during the pack will bump the
reference count for you.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This makes the AGI AsyncAGI event put provided AGI command arguments in
the event's environment.
(closes issue ASTERISK-21304)
Patch-By: Dirk Wendland
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
People who use the features.conf.sample file from Asterisk 11 and before in trunk were
given a rude awakening when features configuration changes were made. Because it uses the
config framework and the config framework is strict about what is accepted and what isn't,
people that had parking options configured found that Asterisk no longer started. This is
because parking options are currently handled in res_parking.conf instead of features.conf.
This fix seeks to create a temporary band-aid fix for the problem, but having parking options
from the general section be passed to a handler that will simply print that the option is no
longer supported. This will not cause Asterisk to exit.
The fix only applies to options in the general section. There are two main reasons for this:
1) The sample features.conf file only has parking options in the general section. There are no
configured parking lots. Therefore it's not quite as "urgent" to get the parking lot parsing
fixed.
2) The plan is to move parking configuration back from res_parking.conf to features.conf. When
that happens, the parking lots will also be addressed at that time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r386792, the ability to play prompts to the first caller in a call queue was
added. While this is arguably a bug fix for those who expect the first caller
to continue receiving prompts while the agent is dialed, it has the side effect
of preventing the first caller from hearing the agent immediately upon
bridging. This may not be a problem for those who really want this option, but
for those who didn't care whether or not the first caller in queue heard their
position, it was an issue.
This patch disables the ability for the first caller in the queue to hear
prompts and adds a new option, announce-to-first-user, to queues.conf. Those
who the behavior can enable it by setting this value to True.
Note that if we ever implement the ability to have the prompts be stopped
upon bridging, this option can be removed.
(closes issue ASTERISK-21782)
Reported by: Remi Quezada
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391245 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds support for Stasis applications to receive bridge-related
messages when the application shows interest in a given bridge.
To supplement this work and test it, this also adds support for the
following bridge-related Stasis-HTTP functionality:
* GET stasis/bridges
* GET stasis/bridges/{bridgeId}
* POST stasis/bridges
* DELETE stasis/bridges/{bridgeId}
* POST stasis/bridges/{bridgeId}/addChannel
* POST stasis/bridges/{bridgeId}/removeChannel
Review: https://reviewboard.asterisk.org/r/2572/
(closes issue ASTERISK-21711)
(closes issue ASTERISK-21621)
(closes issue ASTERISK-21622)
(closes issue ASTERISK-21623)
(closes issue ASTERISK-21624)
(closes issue ASTERISK-21625)
(closes issue ASTERISK-21626)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1). When touching the bridgecallno, we need to lock it.
2). stop_stuff() which calls iax2_destroy_helper()
Assumes the lock on the pvt is already held, when iax2_destroy_helper() is called.
Thus we need to lock the bridgecallno pvt before we call stop_stuff(iaxs[fr->callno]->bridgecallno);
3). When evaluating the state of 'callno->transferring' of the current leg,
we can't change it to READY unless the bridgecallno is locked.
Why, if we are interrupted by the other call leg before 'transferring = TRANSFER_RELEASED',
the interrupt will find that it is READY and that the bridgecallno is also READY so Releases the legs.
(closes issue ASTERISK-21409)
Reported by: alecdavis
Tested by: alecdavis
alecdavis (license 585)
Review https://reviewboard.asterisk.org/r/2594/
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Merged revisions 391063 from http://svn.asterisk.org/svn/asterisk/branches/11
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Topics need to be disposed of prior to the message types that are published
on them. This includes topic pools. This prevents an assertion from being
raised on shutdown.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This moves the initialization call behind the protection against
reloads. We don't want to re-add message router routes during
reloads.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows astmm to access the backtrace generation code in Asterisk.
When memory is allocated, a backtrace is created and stored with the memory
region that tracks the allocation. If a memory corruption is detected, the
backtrace is printed to the astmm log. The backtrace will make use of the
BETTER_BACKTRACES build option if available.
As a result, this patch moves the backtrace generation code into its own file
and uses the non-wrapped versions of the C library memory allocation routines.
This allows the memory allocation code to safely use the backtrace generation
routines without infinitely recursing.
Review: https://reviewboard.asterisk.org/r/2567
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added a start technology callback that technologies can use to start
bridging operations. It is expected that native bridges will find this
useful.
* Factored out bridge_channel_complete_join().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A three party bridge uses the softmix bridging technology. This
technology has a dedicated thread used to perform the analog mixing. When
one of these parties leaves the bridge, the bridge technology is changed
from the softmix technology to a two-party mixing technology. Changing
technologies is done by removing channels from the old technology and
adding them to the new technology. Since the remaining channels do not
leave the bridge, the softmix mixing thread could continue to process all
channels in the bridge. If the bridge code is not able to start
destruction of the softmix technology before the softmix mixing thread
wakes up, a crash happens.
* Added a stop technology callback that technologies can use to request
any helper threads to stop in preparation for being destroyed.
(closes issue AST-1156)
Reported by: John Bigelow
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Stasis cache clear message payloads now consist of a stasis_message
representative of the message to be cleared from the cache. This allows
multiple parallel caches to coexist and be cleared properly by the same
cache clear message even when keyed on different fields.
This change fixes a bug where multiple cache clears could be posted for
channels. The cache clear is now produced in the destructor instead of
ast_hangup.
Additionally, dummy channels are no longer capable of producing channel
snapshots.
Review: https://reviewboard.asterisk.org/r/2596
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Change applicationmap and featuregroup to replace duplicate config items
rather than reject them.
* Remove some unneeded warning messages when getting the applicationmap
allows duplicates from DYNAMIC_FEATURES.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When reading from a config file, it's important to reject duplicates. Otherwise,
featuregroups will have ambiguity when pointing to applicationmap items. However,
when constructing the channel's current applicationmap, we don't care about duplicate
names since it's the DTMF that identifies a feature, not the name.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The channel variable ATTENDED_TRANSFER_COMPLETE_SOUND is no longer
channel driver specific. If the channel variable is set on the
transferrer channel, the sound will be played to the target of an attended
transfer.
* The channel variable BRIDGEPEER becomes a comma separated list of peers
in a multi-party bridge. The BRIDGEPEER value can have a maximum of 10
peers listed. Any more peers in the bridge will not be included in the
list. BRIDGEPEER is not valid in holding bridges like parking since those
channels do not talk to each other even though they are in a bridge.
* The channel variable BRIDGEPVTCALLID is only valid for two party bridges
and will contain a value if the BRIDGEPEER's channel driver supports it.
* The channel variable DYNAMIC_PEERNAME is redundant with BRIDGEPEER and
is removed. The more useful DYNAMIC_WHO_ACTIVATED gives the channel name
that activated the dynamic feature.
* The channel variables DYNAMIC_FEATURENAME and DYNAMIC_WHO_ACTIVATED are
set only on the channel executing the dynamic feature. Executing a
dynamic feature on the bridge peer in a multi-party bridge will execute it
on all peers of the activating channel.
(closes issue ASTERISK-21555)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2582/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Features configuration is handled in its own API in
features_config.h and features_config.c. This way, features
configuration is accessible to anything that needs it.
In addition, features configuration has been altered to
be more channel-oriented. Most callers of features API
code will be supplying a channel so that the individual
channel's settings will be acquired rather than the global
setting.
Missing from this commit is XML documentation for the
features configuration. That will be handled in a separate
commit.
Review: https://reviewboard.asterisk.org/r/2578/
(issue ASTERISK-21542)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix external attended transfer bridge move/swap method. One of the
transferrer channels was not kicked out of the bridge.
* Fix several off-nominal extended attended transfer paths. Mainly the
channels involved needed to be hung up or kicked out of the bridge.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
David M. Lee
Matthew Jordan
Mark Michelson
Jonathan Rose
Igor Goncharovskiy
Alec L Davis
Damien Wedhorn
Kinsey Moore
Richard Mudgett
Jason Parker