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r378374 | rmudgett | 2013-01-02 15:23:16 -0600 (Wed, 02 Jan 2013) | 33 lines
Fix AMI redirect action with two channels failing to redirect both channels.
The AMI redirect action can fail to redirect two channels that are bridged
together. There is a race between the AMI thread redirecting the two
channels and the bridge thread noticing that a channel is hungup from the
redirects.
* Made the bridge wait for both channels to be redirected before exiting.
* Made the AMI redirect check that all required headers are present before
proceeding with the redirection.
* Made the AMI redirect require that any supplied ExtraChannel exist
before proceeding. Previously the code fell back to a single channel
redirect operation.
(closes issue ASTERISK-18975)
Reported by: Ben Klang
(closes issue ASTERISK-19948)
Reported by: Brent Dalgleish
Patches:
jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode
Review: https://reviewboard.asterisk.org/r/2243/
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r378377 | mjordan | 2013-01-02 16:10:32 -0600 (Wed, 02 Jan 2013) | 24 lines
Prevent crashes from occurring when reading from data sources with large values
When reading configuration data from an Asterisk .conf file or when pulling
data from an Asterisk RealTime backend, Asterisk was copying the data on the
stack for manipulation. Unfortunately, it is possible to read configuration
data or realtime data from some data source that provides a large blob of
characters. This could potentially cause a crash via a stack overflow.
This patch prevents large sets of data from being read from an ARA backend or
from an Asterisk conf file.
(issue ASTERISK-20658)
Reported by: wdoekes
Tested by: wdoekes, mmichelson
patches:
* issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674)
* issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674)
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r378384 | mjordan | 2013-01-02 16:19:32 -0600 (Wed, 02 Jan 2013) | 11 lines
Clean up app_mysql's application entry points to properly parse arguments
When parsing arguments, application entry points should not attempt to
directly modify the parameters to the function. This patch properly duplicates
the passed in parameters before attempting to parse them.
(issue ASTERISK-20658)
Reported by: wdoekes
patches:
issueA20658_sanitize_app_mysql.patch uploaded by wdoekes (license 5674)
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r378322 | mjordan | 2013-01-02 12:11:59 -0600 (Wed, 02 Jan 2013) | 33 lines
Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The
device state cache holds the state of each device known to Asterisk, such that
consumers of device state information can query for the last known state for
a particular device, even if it is not part of an active call. The concept of
a device in Asterisk can include entities that do not have a physical
representation. One way that this occurred was when anonymous calls are allowed
in Asterisk. A device was automatically created and stored in the cache for
each anonymous call that occurred; this was possible in the SIP and IAX2
channel drivers and through channel drivers that utilized the
res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices
are never removed from the system, allowing anonymous calls to potentially
exhaust a system's resources.
This patch changes the event cache subsystem and device state management to
no longer cache devices that are not associated with a physical entity.
(issue ASTERISK-20175)
Reported by: Russell Bryant, Leif Madsen, Joshua Colp
Tested by: kmoore
patches:
event-cachability-3.diff uploaded by jcolp (license 5000)
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r378063 | jrose | 2012-12-14 16:34:18 -0600 (Fri, 14 Dec 2012) | 8 lines
Features: BRIDGE_FEATURES variable automixmonitor support and use proper party
BRIDGE_FEATURES did not previously support the automixmonitor feature. Now it
does. In addition, the BRIDGE_FEATURES variable would not apply features to
the proper party based on whether the feature option letter was in caps or
in lowercase (both ways would apply it to the caller). Now uppercase applies
to the caller while lowercase applies to the callee (like with the dial option)
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r378064 | rmudgett | 2012-12-14 16:45:03 -0600 (Fri, 14 Dec 2012) | 4 lines
chan_agent: Remove some duplicated code.
No need to check for an agent twice. Santa does that.
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r377915 | elguero | 2012-12-11 22:43:18 -0600 (Tue, 11 Dec 2012) | 17 lines
Convert Dynamic Features Buffer To Use ast_str
Currently, the buffer for the dynamic features list is set to a fixed size of
128. If the list is bigger than that, it results in the dynamic feature(s) not
being recognized.
This patch changes the buffer from a fixed size to a dynamic one.
(closes issue ASTERISK-20680)
Reported by: Clod Patry
Tested by: Michael L. Young
Patches:
asterisk-20680-dynamic-features-v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2221/
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r377906 | mmichelson | 2012-12-11 16:42:11 -0600 (Tue, 11 Dec 2012) | 3 lines
Add test events necessary for bridging tests to be able to properly run.
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Update and extend the configuration_file group and enable linking. Commit other cleanups from multi-version Doxygen testing. Update title that was left behind many years ago.
(issue ASTERISK-20259)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* ASTERISK-20383
Missing named call pickup group features:
CHANNEL(callgroup) - Need CHANNEL(namedcallgroup)
CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup)
Pickup() - Needs to also select from named pickup groups.
* ASTERISK-20384
Using the pickupexten, the pickup channel selection could fail even though
there was a call it could have picked up. In a call pickup race when
there are multiple calls to pickup and two extensions try to pickup a
call, it is conceivable that the loser will not pick up any call even
though it could have picked up the next oldest matching call.
Regression because of the named call pickup group feature.
* See ASTERISK-20386 for the implementation improvements. These are the
changes in channel.c and channel.h.
* Fixed some locking issues in CHANNEL().
(closes issue ASTERISK-20383)
Reported by: rmudgett
(closes issue ASTERISK-20384)
Reported by: rmudgett
(closes issue ASTERISK-20386)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2112/
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This patch adds the feature "Private representation of caller, connected
and redirecting party ids", as previously discussed with us (DATUS) and
Digium.
1. Feature motivation
Until now it is quite difficult to modify a party number or name which can
only be seen by exactly one particular instantiated technology channel
subscriber. One example where a modified party number or name on one
channel is spread over several channels are supplementary services like
call transfer or pickup. To implement these features Asterisk internally
copies caller and connected ids from one channel to another. Another
example are extension subscriptions. The monitoring entities (watchers)
are notified of state changes and - if desired - of party numbers or names
which represent the involving call parties. One major feature where a
private representation of party names is essentially needed, i.e. where a
party name shall be exclusively signaled to only one particular user, is a
private user-specific name resolution for party numbers. A lookup in a
private destination-dependent telephone book shall provide party names
which cannot be seen by any other user at any time.
2. Feature Description
This feature comes along with the implementation of additional private
party id elements for caller id, connected id and redirecting ids inside
Asterisk channels.
The private party id elements can be read or set by the user using
Asterisk dialplan functions.
When a technology channel is initiating a call, receives an internal
connected-line update event, or receives an internal redirecting update
event, it merges the corresponding public id with the private id to create
an effective party id. The effective party id is then used for protocol
signaling.
The channel technologies which initially support the private id
representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and
PRI (chan_dahdi).
Once a private name or number on a channel is set and (implicitly) made
valid, it is generally used for any further protocol signaling until it is
rewritten or invalidated.
To simplify the invalidation of private ids all internally generated
connected/redirecting update events and also all connected/redirecting
update events which are generated by technology channels -- receiving
regarding protocol information - automatically trigger the invalidation of
private ids.
If not using the private party id representation feature at all, i.e. if
using only the 'regular' caller-id, connected and redirecting related
functions, the current characteristic of Asterisk is not affected by the
new extended functionality.
3. User interface Description
To grant access to the private name and number representation from the
Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan
functions are extended by the following data types. The formats of these
data types are equal to the corresponding regular 'non-private' already
existing data types:
CALLERID:
priv-all
priv-name priv-name-valid priv-name-charset priv-name-pres
priv-num priv-num-valid priv-num-plan priv-num-pres
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag
CONNECTEDLINE:
priv-name priv-name-valid priv-name-pres priv-name-charset
priv-num priv-num-valid priv-num-pres priv-num-plan
priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd
priv-tag
REDIRECTING:
priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset
priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd
priv-orig-tag
priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset
priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan
priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd
priv-from-tag
priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset
priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan
priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
priv-to-tag
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2030/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
You cannot unref a pointer and then expect to ref it again later.
* Fix potential NULL pointer deref if the call pickup search fails.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are
implemented in parallel to the existing numbered callgroup/pickupgroup
implementation. However, unlike the existing implementation, which is limited
to a maximum of 64 defined groups, the number of defined groups allowed for
named callgroups/pickupgroups is effectively unlimited.
Named groups are configured with the keywords "namedcallgroup" and
"namedpickupgroup". This corresponds to the numbered group definitions of
"callgroup" and "pickupgroup". Note that as the implementation of named groups
coexists with the existing numbered implementation, a defined named group of
"4" does not equate to numbered group 4.
Support for the named groups has been added to the SIP, DAHDI, and mISDN channel
drivers.
Review: https://reviewboard.asterisk.org/r/2043
Uploaded by:
Guenther Kelleter(license #6372)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Because some of the manager events are defined in the top of the source, due
to the macro calls not containing all necessary information to have the
documentation colocated with the call itself, several include statements were
failing when built with 'make'. While this did not cause any problems in
compilation or validation, it did result in a number of warnings being dumped
to stderr.
This patch changes those references such that they always resolve, regardless
of the documentation build options.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds some basic documentation for a number of modules. This
includes core source files in Asterisk (those in main), as well as
chan_agent, chan_dahdi, chan_local, sig_analog, and sig_pri. The DTD
has also been updated to allow referencing of AMI commands.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Hangup handlers are an alternative to the h extension. They can be used
in addition to the h extension. The idea is to attach a Gosub routine to
a channel that will execute when the call hangs up. Whereas which h
extension gets executed depends on the location of dialplan execution when
the call hangs up, hangup handlers are attached to the call channel. You
can attach multiple handlers that will execute in the order of most
recently added first.
(closes issue ASTERISK-19549)
Reported by: Mark Murawski
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/2002/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Adds call ID logging changes to specific channel drivers that weren't handled
handled in phase II of Call ID Logging. Also covers logging for threads for
threads created by systems that may be involved with many different calls.
Extra special thanks to Richard for rigorous review of chan_dahdi and its
various signalling modules.
review: https://reviewboard.asterisk.org/r/1927/
review: https://reviewboard.asterisk.org/r/1950/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix do_bridge_masquerade() getting the resume location from the zombie
channel. The code must not touch a clone channel after it has masqueraded
it. The clone channel has become a zombie and is starting to hangup.
(closes issue ASTERISK-19985)
Reported by: jamicque
Patches:
jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: jamicque
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* Fix AMI Bridge action disconnecting the AMI link on error.
* Fix AMI Bridge action and Bridge application not checking if their
masquerades were successful.
* Fix Bridge application running the h-exten when it should not.
* Made do_bridge_masquerade() return if the masquerade was successful so
the Bridge application and AMI Bridge action could deal with it correctly.
* Made bridge_call_thread_launch() hangup the passed in channels if the
bridge_call_thread fails to start. Those channels would have been
orphaned.
* Made builtin_atxfer() check the success of the transfer masquerade
setup.
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When the IAX2 Who Hung Up? changes were added, they uncovered a bug in
the way AST_CONTROL_PVT_CAUSE_CODE was handled in
feature_request_and_dial(). This particular frame subtype was being
treated like more terminal control frames causing the function to be
exited prematurely.
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This was essentially duplicated functionality where normal channels used
AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used
AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts that usage
into AST_CAUSE_ANSWERED_ELSEWHER usage.
Review: https://reviewboard.asterisk.org/r/1944
(closes issue ASTERISK-19865)
Patch-by: Birger Harzenetter
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Presence support has been added. This is accomplished by
allowing for presence hints in addition to device state
hints. A dialplan function called PRESENCE_STATE has been
added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML
elements in a PIDF presence document.
Voicemail has new APIs that allow for moving, removing,
forwarding, and playing messages. Messages have had a new
unique message ID added to them so that the APIs will work
reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox.
A voicemail Dialplan App called VoiceMailPlayMsg has been
added to be able to play back a specific message.
Configuration hooks have been added. Configuration hooks
allow for a piece of code to be executed when a specific
configuration file is loaded by a specific module. This is
useful for modules that are dependent on the configuration
of other modules.
chan_sip now has a public method that allows for a custom
SIP INFO request to be sent mid-dialog. Digium phones use
this in order to display progress bars when files are played.
Messaging support has been expanded a bit. The main
visible difference is the addition of an AMI action
MessageSend.
Finally, a ParkingLots manager action has been added in order
to get a list of parking lots.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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
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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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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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
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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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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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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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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Automerge script
Richard Mudgett
Andrew Latham
Matthew Jordan
Mark Michelson
Kinsey Moore
Jonathan Rose
Russell Bryant
Terry Wilson