When remotely bridging calls with directmedia, Asterisk would check
the address of the peers/users holding directmedia ACLs (set via
directmediapermit/directmediadeny) instead of the bridged peer. This
is similar to r366547, but trunk specific and involves changes to
the rtpengine instead of just chan_sip.
(closes issue AST-876)
review: https://reviewboard.asterisk.org/r/1924/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Uncovered a nasty reference leak while I was writing some changes to
chan_dahdi/sig_analog. Slapped myself around a bit after seeing that I
performed the unchecked return causing this problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch fixes to situations that could cause the CEL LINKEDID_END event to
be missed.
1) During a core stop gracefully, modules are unloaded when ast_active_channels
== 0. The LINKDEDID_END event fires during the channel destructor. This means
that occasionally, the cel_* module will be unloaded before the channel is
destroyed. It seemed generally useful to wait until the refcount of all
channels == 0 before unloading, so I added a channel counter and used it in the
shutdown code.
2) During a masquerade, ast_channel_change_linkedid is called. It calls
ast_cel_check_retire_linkedid which unrefs the linkedid in the linkedids
container in cel.c. It didn't ref the new linkedid. Now it does.
Review: https://reviewboard.asterisk.org/r/1900/
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This change accommodates two methods by which calls can be directed to
a user's voicemail.
* Incoming calls can be redirected to any user's voicemail.
* Established calls can be blind transferred to any user's voicemail.
Digium phones indicate the desire to direct a call to voicemail by using
a Diversion header with a reason parameter of "send_to_vm".
This patch adds the "send_to_vm" reason as a valid redirecting reason. In
addition, chan_sip.c has been modified to update redirecting information
on the transferred channel by reading a Diversion header on a REFER request.
(closes issue AST-871)
Reported by Malcolm Davenport
Review: https://reviewboard.asterisk.org/r/1925
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This addresses core findings 4 and 6.
Moises Silva helped me by stating that a break could be
safely added to the case where it is added in chan_dahdi.c
In say.c, I have added a comment indicating that static analysis
complains but that it is currently unknown if this is correct.
This fixes all core findings of this type.
(closes issue ASTERISK-19662)
reported by Matthew Jordan
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SSL_CTX structures were allocated but never freed. This was a bigger
issue for clients than servers since new SSL_CTX structures could be
allocated for each connection. Servers, on the other hand, typically
set up a single SSL_CTX for their lifetime.
This is solved in two ways:
1. In __ssl_setup(), if a tcptls_cfg has an ssl_ctx on it, it is
freed so that a new one can take its place.
2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
been added so that servers can properly free their SSL_CTXs.
(issue ASTERISK-19278)
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This patch adds to what was fixed in r366880. Specifically, it addresses the
following:
* chan_sip: dispose of an allocated frame in off nominal code paths in
sip_rtp_read
* func_odbc: when disposing of an allocated resultset, ensure that any rows
that were appended to that resultset are also disposed of
* cli: free the created return string buffer in another off nominal code
path
* chan_dahdi: free a frame that was allocated by the dsp layer if we choose
not to process that frame
(issue ASTERISK-19665)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922/
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This patch addresses a number of memory leaks in a variety of modules that were
found by a static analysis tool. A brief summary of the changes:
* app_minivm: free ast_str objects on off nominal paths
* app_page: free the ast_dial object if the requested channel technology
cannot be appended to the dialing structure
* app_queue: if a penalty rule failed to match any existing rule list
names, the created rule would not be inserted and its memory
would be leaked
* app_read: dispose of the created silence detector in the presence of
off nominal circumstances
* app_voicemail: dispose of an allocated unique ID field for MWI event
un-subscribe requests in off nominal paths; dispose of
configuration objects when using the secret.conf option
* chan_dahdi: dispose of the allocated frame produced by ast_dsp_process
* chan_iax2: properly unref peer in CLI command "iax2 unregister"
* chan_sip: dispose of the allocated frame produced by sip_rtp_read's
call of ast_dsp_process; free memory in parse unit tests
* func_dialgroup: properly deref ao2 object grhead in nominal path of
dialgroup_read
* func_odbc: free resultset in off nominal paths of odbc_read
* cli: free match_list in off nominal paths of CLI match completion
* config: free comment_buffer/list_buffer when configuration file load
is unchanged; free the same buffers any time they were
created and config files were processed
* data: free XML nodes in various places
* enum: free context buffer in off nominal paths
* features: free ast_call_feature in off nominal paths of applicationmap
config processing
* netsock2: users of ast_sockaddr_resolve pass in an ast_sockaddr struct
that is allocated by the method. Failures in
ast_sockaddr_resolve could result in the users of the method
not knowing whether or not the buffer was allocated. The
method will now not allocate the ast_sockaddr struct if it
will return failure.
* pbx: cleanup hash table traversals in off nominal paths; free
ignore pattern buffer if it already exists for the specified
context
* xmldoc: cleanup various nodes when we no longer need them
* main/editline: various cleanup of pointers not being freed before being
assigned to other memory, cleanup along off nominal paths
* menuselect/mxml: cleanup of value buffer for an attribute when that attribute
did not specify a value
* res_calendar*: responses are allocated via the various *_request method
returns and should not be allocated in the various
write_event methods; ensure attendee buffer is freed if no
data exists in the parsed node; ensure that calendar objects
are de-ref'd appropriately
* res_jabber: free buffer in off nominal path
* res_musiconhold: close the DIR* object in off nominal paths
* res_rtp_asterisk: if we run out of ports, close the rtp socket object and free
the rtp object
* res_srtp: if we fail to create the session in libsrtp, destroy the
temporary ast_srtp object
(issue ASTERISK-19665)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922
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There was a slight discrepancy in the behaviors of the old SIP_CAUSE and the
new SIP_CAUSE/HANGUPCAUSE when a channel had been originated and had not yet
been answered. This caused the noload_res_srtp_attempt_srtp test to fail since
the SIP_CAUSE variable was never actually set. This behavior has been restored.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch improves the handling of call id logging significantly with regard
to transfers and adding APIs to better handle specific aspects of logging.
Also, changes have been made to chan_sip in order to better handle the creation
of callids and to enable the monitor thread to bind itself to a particular
call id when a dialog is determined to be related to a callid. It then unbinds
itself before returning to normal monitoring.
review: https://reviewboard.asterisk.org/r/1886/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366842 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the starting point for the Asterisk 11: Who Hung Up work and provides
a framework which will allow channel drivers to report the types of hangup
cause information available in SIP_CAUSE without incurring the overhead of the
MASTER_CHANNEL dialplan function. The initial implementation only includes
cause generation for chan_sip and does not include cause code translation
utilities.
This change deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the 'storesipcause'
option in sip.conf.
Review: https://reviewboard.asterisk.org/r/1822/
(Closes issue SWP-4221)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Before this patch, the predial routine executes on the ;1 channel of a
local channel pair. Executing predial on the ;1 channel of a local
channel pair is of limited utility. Any channel variables set by the
predial routine executing on the ;1 channel will not be available when the
local channel executes dialplan on the ;2 channel.
* Create ast_pre_call() and an associated pre_call() technology callback
to handle running the predial routine. If a channel technology does not
provide the callback, the predial routine is simply run on the channel.
Review: https://reviewboard.asterisk.org/r/1903/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20,
22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111,
and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped
were either extended/deprecated or in areas of code that shouldn't be
disturbed.
(Closes issue ASTERISK-19650)
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The VoiceMail d([c]) option is documented to accept digits for a new extension
in context <c>, if played during the greeting. This option works fine if the
extension being redirected to has an extension with the same initial digit in
the channel's current context. If that digit did not happen to exist in some
extension, a dialplan match would fail and the user would not be redirected.
This patch fixes it such that if the <c> option is used, the extensions are
matched in that context as opposed to the caller's original context.
(closes issue ASTERISK-18243)
Reported by: mjordan
Tested by: mjordan
Review: https://reviewboard.asterisk.org/r/1892
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Most of the changes here are trivial NULL checks. There are a couple
optimizations to remove the need to check for NULL and outboundproxy parsing
in chan_sip.c was rewritten to avoid use of strtok. Additionally, a bug was
found and fixed with the parsing of outboundproxy when "outboundproxy=," was
set.
(Closes issue ASTERISK-19654)
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r365006 | twilson | 2012-05-02 10:49:03 -0500 (Wed, 02 May 2012) | 12 lines
Fix a CEL LINKEDID_END race and local channel linkedids
This patch has the ;2 channel inherit the linkedid of the ;1 channel and fixes
the race condition by no longer scanning the channel list for "other" channels
with the same linkedid. Instead, cel.c has an ao2 container of linkedid strings
and uses the refcount of the string as a counter of how many channels with the
linkedid exist. Not only does this eliminate the race condition, but it also
allows us to look up the linkedid by the hashed key instead of traversing the
entire channel list.
Review: https://reviewboard.asterisk.org/r/1895/
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r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 May 2012) | 11 lines
Don't leak a ref if out of memory and can't link the linkedid
If the ao2_link fails, we are most likely out of memory and bad things
are going to happen. Before those bad things happen, make sure to clean
up the linkedid references.
This patch also adds a comment explaining why linkedid can't be passed
to both local channel allocations and combines two ao2_ref calls into 1.
Review: https://reviewboard.asterisk.org/r/1895/
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In audiohook_read_frame_both, anytime samples are obtained from the read/write
factories a debug statement is logged stating that samples were not obtained
from the factories. This statement used to only occur if option_debug was
turned on and no samples were obtained; in some refactoring when the
option_debug statement was removed, the "else" clause was removed as well.
This patch makes it so that those debug log statements only occur if the
condition leading up to them actually happened.
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md5.c: In function ‘MD5Final’:
md5.c:154:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing]
md5.c:155:2: error: dereferencing type-punned pointer will break strict-aliasing rules [-Werror=strict-aliasing]
There is an md5 unit test and it still passes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add two new dialplan functions: FEATURE() and FEATUREMAP(). FEATURE()
lets you set some of the configuration options from the [general] section
of features.conf on a per-channel basis. FEATUREMAP() lets you customize
the key sequence used to activate built-in features, such as blindxfer,
and automon. See the built-in documentation for details.
Review: https://reviewboard.asterisk.org/r/1871/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r364365 | twilson | 2012-04-27 17:31:01 -0500 (Fri, 27 Apr 2012) | 11 lines
Fix ast_parse_arg numeric type range checking and add tests
ast_parse_arg wasn't checking for strto* parse errors or limiting
the results by the actual range of the numeric types. This patch fixes
that and adds unit tests as well.
Review: https://reviewboard.asterisk.org/r/1879/
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r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) | 2 lines
Add missing test_config.c
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When party B does an attended transfer of party A to party C, the
attending bridge between party B and C should not be running an h exten
when the bridge ends. Running an h exten now sets a softhangup flag to
ensure that an AGI will run in dead AGI mode.
* Set the AST_FLAG_BRIDGE_HANGUP_DONT on the party B channel for the
attending bridge between party B and C.
(closes issue AST-870)
(closes issue ASTERISK-19717)
Reported by: Mario
(closes issue ASTERISK-19633)
Reported by: Andrey Solovyev
Patches:
jira_asterisk_19633_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Andrey Solovyev, Mario
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Asterisk has a setting for the minimum allowed DTMF. If we get shorter
DTMF tones, these will be changed to the minimum on the outbound call
leg.
(closes issue ASTERISK-19772)
Review: https://reviewboard.asterisk.org/r/1882/
Reported by: oej
Tested by: oej
Patches by: oej
Thanks to the reviewers.
1.8 branch for this patch: agave-dtmf-duration-asterisk-conf-1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Found a small amount of curly brackets in my hotel room here in Denmark.
I hereby donate them to the Asterisk project.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) B calls A with Dial option T
2) B DTMF atxfer to C
3) B hangs up
4) C does not answer
5) B is called back
6) B answers
7) B cannot initiate transfers anymore
* Add dial features datastore to recalled party B channel that is a copy
of the original party B channel's dial features datastore.
* Extracted add_features_datastore() from add_features_datastores().
* Renamed struct ast_dial_features features_caller and features_callee
members to my_features and peer_features respectively. These better names
eliminate the need for some explanatory comments.
* Simplified code accessing the struct ast_dial_features datastore.
(closes issue ASTERISK-19383)
Reported by: lgfsantos
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As detailed in the advisory, AMI users without write authorization for SYSTEM class AMI
actions were able to run system commands by going through other AMI commands which did
not require that authorization. Specifically, GetVar and Status allowed users to do this
by setting their variable/s options to the SHELL or EVAL functions.
Also, within 1.8, 10, and trunk there was a similar flaw with the Originate action that
allowed users with originate permission to run MixMonitor and supply a shell command
in the Data argument. That flaw is fixed in those versions of this patch.
(closes issue ASTERISK-17465)
Reported By: David Woolley
Patches:
162_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
18_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
10_ami_readfunc_security_r2.diff uploaded by jrose (license 6182)
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* Redo ast_app_run_sub()/ast_app_exec_sub() to use a known return point so
execution will stop after the routine returns there.
(s@gosub_virtual_context:1)
* Create ast_app_exec_macro() and ast_app_exec_sub() to run the macro and
gosub application respectively with the parameter string already created.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Move debug message in ast_rtp_instance_early_bridge_make_compatible() to
be output when what it states has actually happened.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Security Events Framework API was changed while adding the generation of
security events in chan_sip. A payload type and name was missed from being
added to struct ie_maps.
(closes issue ASTERISK-19759)
Reported by: Michael L. Young
Patches:
issue-asterisk-19759.diff uploaded by Michael L. Young (license 5026)
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The CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the channel
lock was originally obtained is overkill where ast_channel_lock_both() was
inlined.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ISDN ETSI PTP and Q.SIG (And SS7 in future) have support for reporting who
was the original redirecting party of a call.
* Added support for the original redirecting party and reason to the
REDIRECTING function and the system core as well as to the stubbed
locations in sig_pri.c.
Review: https://reviewboard.asterisk.org/r/1829/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* chan_mobile: Fixed an overrun where the cind_state buffer (an integer array
of size 16) would be overrun due to improper bounds checking. At worst, the
buffer can be overrun by a total of 48 bytes (assuming 4-byte integers),
which would still leave it within the allocated memory of struct hfp. This
would corrupt other elements in that struct but not necessarily cause any
further issues.
* app_sms: The array imsg is of size 250, while the array (ud) that the data
is copied into is of size 160. If the size of the inbound message is
greater then 160, up to 90 bytes could be overrun in ud. This would corrupt
the user data header (array udh) adjacent to ud.
* chan_unistim: A number of invalid memmoves are corrected. These would move
data (which may or may not be valid) into the ends of these buffers.
* asterisk: ast_console_toggle_loglevel does not check that the console log
level being set is less then or equal to the allowed log levels of 32.
* format_pref: In ast_codec_pref_prepend, if any occurrence of the specified
codec is not found, the value used to index into the array pref->order
would be one greater then the maximum size of the array.
* jitterbuf: If the element being placed into the jitter buffer lands in the
last available slot in the jitter history buffer, the insertion sort attempts
to move the last entry in the buffer into one slot past the maximum length
of the buffer. Note that this occurred for both the min and max jitter
history buffers.
* tdd: If a read from fsk_serial returns a character that is greater then 32,
an attempt to read past one of the statically defined arrays containing the
values that character maps to would occur.
* localtime: struct ast_time and tm are not the same size - ast_time is larger,
although it contains the elements of tm within it in the same layout. Hence,
when using memcpy to copy the contents of tm into ast_time, the size of tm
should be used, as opposed to the size of ast_time.
* extconf: this treats ast_timing's minmask array as if it had a length of 48,
when it has defined the size of the array as 24. pbx.h defines minmask as
having a size of 48.
(issue ASTERISK-19668)
Reported by: Matt Jordan
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This patch addresses a number of modules in main that did not handle the
negative return value from function calls adequately, or were not sufficiently
clear that the conditions leading to improper handling of the return values
could not occur. This includes:
* asterisk.c: A negative return value from the read function would be used
directly as an index into a buffer. We now check for success of the read
function prior to using its result as an index.
* manager.c: Check for failures in mkstemp and lseek when handling the
temporary file created for processing data returned from a CLI command in
action_command. Also check that the result of an lseek is sanitized prior
to using it as the size of a memory map to allocate.
(issue ASTERISK-19655)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1863/
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The current Security Events Framework API only supports IPv4 when it comes to
generating security events. This patch does the following:
* Changes the Security Events Framework API to support IPV6 and updates
the components that use this API.
* Eliminates an error message that was being generated since the current
implementation was treating an IPv6 socket address as if it was IPv4.
* Some copyright dates were updated on files touched by this patch.
(closes issue ASTERISK-19447)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
security_events_ipv6v3.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1777/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ASTERISK-18809 eliminated the legacy macro invocation of the stdexten in
favor of the Gosub method without a means of backwards compatibility.
(issue ASTERISK-18809)
(closes issue ASTERISK-19457)
Reported by: Matt Jordan
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1855/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Doing Set(MESSAGE_DATA(key)=) would add an empty key header if the key
header did not already exist. If it already existed it would delete it.
* Made msg_set_var_full() exit early if the named variable did not already
exist and the value to set is empty.
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* Hangup now can take a regular expression as the Channel option. If you want
to hangup multiple channels, use /regex/ as the Channel option. Existing
behavior to hanging up a single channel is unchanged, but if you pass a regex,
the manager will send you a list of channels back that were hung up.
(closes issue ASTERISK-19575)
Reported by: Mark Murawski
Tested by: Mark Murawski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Prior to this patch, a connected line update was queued during
call pickup and then an answer frame was queued. The original
caller would presumably then have his connected line updated
and then the call would be answered.
In actuality, the answer frame was not how the call ended up
being answered. Rather, an odd section in app_dial that checks
if the called channel's state is up.
The result is that the order of the connected line update and
the answer were variable. In most cases, this wasn't actually
a bad thing. However, if the 'I' option was passed to dial, the
connected line update would be inhibited.
The fix is to queued the connected line after the answer frame is
queued. This way the race in app_dial is between two
conditions resulting in an answer. This way the connected line
update occurs after the answer every time.
(closes issue ASTERISK-19183)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Mark Michelson
Patches:
ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
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* Rename astobj2 API parameter funcname to func.
* Rename astobj2 API iterator parameter to iter.
* Update some documentation for OBJ_MULTIPLE.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Global ao2 objects must always exist after initialization because there is
no access control to obtain another reference to the global object.
It is expected that module configuration could use these new API calls to
replace an active configuration parameter object with an updated
configuration parameter object.
With these new API calls, the global object could be replaced, removed, or
referenced without the risk of someone using a stale global object
pointer.
Review: https://reviewboard.asterisk.org/r/1824/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Rather then flood the CLI with verbose messages, we've changed the level to
debug. This will help keep the CLI clean.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) | 6 lines
expression parser: Fix (theoretical) memory leak.
Fix a memory leak that is very unlikely to actually happen. If a malloc()
succeeded, but the following strdup() failed, the memory from the original
malloc() would be leaked.
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r360357 | russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
Rebuild parsers.
This is needed to include the last fix to main/ast_expr2.y. The changes look
much bigger as this regeneration of the code was done with newer versions of
flex and bison.
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Q.951 indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening indicator
field should be "Network provided".
* Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to interworking". This
fix makes Asterisk consistent and it also makes it consistent with earlier
branches as far as this presentation value is concerned.
* Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions
handle the "Number not available due to interworking" case better in
sig_pri.c. This change is possible because the minimum required libpri
version (v1.4.11) has the necessary defines in libpri.h.
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Everything still compiled after making these changes, so I assume these
whitespace-only changes didn't break anything (and shouldn't have).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fix AMI module reload deadlock regression from ASTERISK-18479 when it
tried to fix the race between calling an AMI action callback and
unregistering that action. Refixes ASTERISK-13784 broken by
ASTERISK-17785 change.
Locking the ao2 object guaranteed that there were no active callbacks that
mattered when ast_manager_unregister() was called. Unfortunately, this
causes the deadlock situation. The patch stops locking the ao2 object to
allow multiple threads to invoke the callback re-entrantly. There is no
way to guarantee a module unload will not crash because of an active
callback. The code attempts to minimize the chance with the registered
flag and the maximum 5 second delay before ast_manager_unregister()
returns.
The trunk version of the patch changes the API to fix the race condition
correctly to prevent the module code from unloading from memory while an
action callback is active.
* Don't hold the lock while calling the AMI action callback.
(closes issue ASTERISK-19487)
Reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1818/
Review: https://reviewboard.asterisk.org/r/1820/
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* Remove unnnecessary const from const char * const var declaration in the
ast_app_run_macro() and ast_app_run_sub() prototypes. The second const is
unnecessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There exists a remotely exploitable stack buffer overflow in HTTP digest
authentication handling in Asterisk. The particular method in question
is only utilized by HTTP AMI. When parsing the digest information, the
length of the string is not checked when it is copied into temporary buffers
allocated on the stack.
This patch fixes this behavior by parsing out pre-defined key/value pairs
and avoiding unnecessary copies to the stack.
(closes issue ASTERISK-19542)
Reported by: Russell Bryant
Tested by: Matt Jordan
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* Added 'b' and 'B' options to Dial. These options will allow you to run
last-minute dialplan on the caller and callee channels while the Dial
application is executing, but before the call is started. For example you
can use the 'b' option to run dialplan on the callee channel to get the name
of the newly created channel right away.
Review: https://reviewboard.asterisk.org/r/1229/
(closes issue: ASTERISK-19548)
Reported by: Mark Murawski
Tested by: Mark Murawski, Stefan Schmidt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a change in time occurs, such that the timestamps associated with frames
being placed into an adaptive jitter buffer (implemented in jitterbuf.c)
are significantly different then the previously inserted frames, the jitter
buffer checks to see if it needs to be resynched to the new time frame. If
three consecutive packets break the threshold, the jitter buffer resynchs
itself to the new timestamps. This currently only occurs when history is
calculated, and hence only on JB_TYPE_VOICE frames.
JB_TYPE_CONTROL frames, on the other hand, are never passed to the history
calculations. Because of this, if the jump in time is greater then the
maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are
dropped and no resynchronization occurs. Alterntively, if the overfill
logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the
buffer, but with a time reference that is not applicable. Subsequent
JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads
from the jitter buffer reach the errant JB_TYPE_CONTROL frame.
This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As
JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the
resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch
threshold.
Note that this only impacts chan_iax2, as other consumers of the adaptive
jitter buffer use the abstract jitter buffer API, which does not use
JB_TYPE_CONTROL frames.
Review: https://reviewboard.asterisk.org/r/1814/
(closes issue ASTERISK-18964)
Reported by: Kris Shaw
Tested by: Kris Shaw, Matt Jordan
Patches:
jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722)
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When connected line support was added, the wait_for_answer() variable
single changed its meaning slightly. Unfortunately, the places where
single was used did not necessarily get updated to reflect that change.
Also audio/video frames were sent to all forked calls when the endpoints
were never made compatible.
* Don't pass audio/video media frames when the channels have not been made
compatible.
* Added handling of AST_CONTROL_SRCCHANGE to app_dial.c.
* Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also
pass a requested MOH class.
(closes issue ASTERISK-16901)
Reported by: Chris Gentle
(closes issue ASTERISK-17541)
Reported by: clint
Review: https://reviewboard.asterisk.org/r/1805/
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This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact
that logger.c implements 32 log levels (because of the custom log level stuff).
asterisk.c uses this define to size an array of levels per remote console.
This array is modified in ast_console_toggle_loglevel(), which is called by the
"logger set level" CLI command. While the documentation for the CLI command
doesn't make it terribly obvious, you can use this CLI command to toggle a
custom log level on a remote console, as well. However, doing so led to an
invalid array index in asterisk.c.
This array is read from any time a log message is written to a console. So,
all custom log level messages resulted in a bogus read if a remote console
was connected.
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The set_format() function was more subtle in how it modified the
struct ast_channel readtrans/writetrans values.
* Fixed ast_activate_generator() conversion correctly.
(closes issue ASTERISK-19434)
Reported by: Birger Harzenetter
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The process_output function in manager.c attempted to call fclose and close immediately
afterwards. Since fclose implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error checking to fclose and
close depending on which was deemed necessary. Also error messages. Thanks to Rosen
Iliev for pointing out the location of the problem.
(closes issue ASTERISK-18453)
Reported By: Jaco Kroon
Review: https://reviewboard.asterisk.org/r/1793/
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This change fixes case-sensitivity for device-specific subscriptions such that
the technology identifier is case-insensitive while the remainder of the device
string is still case-sensitive. This should also preserve the original case of
the device string as passed in to the event system. CCSS is the only feature
affected as it is the only consumer of device-specific event subscriptions.
The second part of this patch addresses similar case-sensitivity issues within
CCSS itself that prevented it from functioning correctly after the fix to the
events system.
This adds a unit test to verify that the event system works as expected.
(closes issue ASTERISK-19422)
Review: https://reviewboard.asterisk.org/r/1780/
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Richard Mudgett
Jonathan Rose
Mark Michelson
Terry Wilson
Matthew Jordan
Kinsey Moore
Jason Parker
Mark Murawki
Russell Bryant
Olle Johansson
Tilghman Lesher
Michael L. Young
Walter Doekes
Paul Belanger
Joshua Colp