A number of applications/AMI commands in Asterisk have specific behavioral
differences depending on the resource or channel technology those
applications are executed on. For example, the MessageSend application/
command is technology agnostic, but how the channel drivers that support
that functionality behave is dependant on the protocols and channel
driver implementation. Prior to this patch, those details were either
documented in the application/command documentation itself, or were left
undocumented.
This patch adds a new element to the documentation schema, <info/>. An info
node is essentially a piece of technology specific reference information that
can be included by any top level XML documentation node. For example, the
MessageSend application can now include XMPP/SIP specific information, where
that technology specific information can be defined in chan_motif/res_xmpp/
chan_sip. Likewise, that information can also be included in the MessageSend
AMI command.
Review: https://reviewboard.asterisk.org/r/2049
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
To fix a memory leak in CEL, a channel datastore was introduced whose
destruction function pointer was pointed to the ast_free macro. Without
MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free.
With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a
different place then utils.h, and became undefined. This patch resolves this
by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be
ast_free, which is defined to be free.
(issue AST-916)
Reported by: Thomas Arimont
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A customer reported a significant memory leak using Asterisk 1.8. They
have tracked it down to ast_cel_fabricate_channel_from_event() in
main/cel.c, which is called by both in-tree CEL logging modules
(cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event
that they log.
The cause was an incorrect assumption about how data attached to an
ast_channel would be handled when the channel is destroyed; the data
is now stored in a datastore attached to the channel, which is
destroyed along with the channel at the proper time.
(closes issue AST-916)
Reported by: Thomas Arimont
Review: https://reviewboard.asterisk.org/r/2053/
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This patch adds Named ACL functionality to Asterisk. This allows system
administrators to define an ACL and refer to it by a unique name. Configurable
items can then refer to that name when specifying access control lists.
It also includes updates to all core supported consumers of ACLs. That includes
manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk
by Olle E. Johansson and provides a subset of the Named ACL functionality
implemented in that branch. For more information on this feature, see acl.conf
and/or the Asterisk wiki.
Review: https://reviewboard.asterisk.org/r/1978/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 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
The awk script parses out the first instance of the DOCUMENTATION tag that it
finds within a file. If a file did not previously have a DOCUMENTATION tag
but received one due to it having an AMI event, then the XML fragment
associated with the AMI event was erroneously placed in the resulting XML
file. Without the python scripts, these XML fragments will not validate.
This patch adds DOCUMENTATION tags at the top of those files that did
not previously have them to prevent the awk script from pulling AMI event
documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369910 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
When the OpenSSL duplicate initialization issues were resolved in r351447,
res_curl could fail to load if it checked SSL_library_init after SSL
initialization completed. This is due to the SSL_library_init stub returning
a value of 0 for success, as opposed to a value of 1. OpenSSL uses a value of
1 to indicate success - in fact, SSL_library_init is documented to always return
1. Interestingly, the CURL libraries actually checked the return value - the fact
that nothing else that depends on OpenSSL was having problems loading probably means
they don't check the return value.
(closes issue AST-924)
Reported by: Guenther Kelleter
patches:
(AST-924.patch license #6372 uploaded by Guenther Kelleter)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369870 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Failure to do this results in the Google Talk client ignoring RTP packets after a specific period of time. This is also done as a result of receiving a STUN binding request so that the username information can be used from the inbound request, thus not requiring it to be stored on a per candidate basis.
(closes issue ASTERISK-20107)
Reported by: Malcolm Davenport
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The problem here is that multiple server sessions share
a SSL_CTX. When one session ended, the SSL_CTX would be
freed and set NULL, leaving the other sessions unable to
function.
The code being removed is superfluous because the SSL_CTX
structures for servers will be properly freed when ast_ssl_teardown
is called.
(closes issue ASTERISK-20074)
Reported by Trevor Helmsley
Patches:
ASTERISK-20074.diff uploaded by Mark Michelson (license #5049)
Testers:
Trevor Helmsley
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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
The opaquing the ast_channel struct no longer requires .cleancount to be
changed when the struct is changed.
* Bump .cleancount value one last time because of struct ast_channel for
old times sake.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Allows the setting of flags via the config options api.
For example, code like this:
#define OPT1 1 << 0
#define OPT2 1 << 1
#define OPT3 1 << 2
struct thing {
unsigned int flags;
};
and a config like this:
[blah]
opt1=yes
opt2=no
opt3=yes
Review: https://reviewboard.asterisk.org/r/2004/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369454 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
When res_adsi is unloaded, it removes the ADSI functions that it previously installed
by passing a NULL adsi_funcs pointer to ast_adsi_install_funcs. This function was not
checking whether or not the adsi_funcs pointer passed in was NULL before dereferencing
it to check whether or not the version of the functions matches what the core was
expecting it.
This patch makes it so that the version is only checked if a potentially valid adsi_funcs
pointer was passed in. Passing in NULL removes the installed functions, bypassing the
version check.
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Thank you rmudgett for pointing out that I was missing this in the initial
check-in for AMI event documentation (r369346)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Certain places in core/cdr.c would, if the duration value were 0, calculate the
duration as being the delta between the current time and the time at which the
CDR record was started. While this does not typically cause a problem in
non-batch mode, this can cause an issue in batch mode where CDR records are
gathered and written long after those calls have ended. In particular, this
affects calls that were never answered, as those are expected to have a duration
of 0. Often, this would result in CDR logs with a significant number of calls
with lengthy durations, but dispositions of "BUSY".
Note that this does not affect cdr_csv, as that backend does not use
ast_cdr_getvar and instead directly reports the duration value. The affected
core backends include cdr_apative_odbc and cdr_custom; other extended or
deprecated CDR backends may potentially still directly manipulate the duration
values.
(issue ASTERISK-19860)
Reported by: Thomas Arimont
(issue AST-883)
Reported by: Thomas Arimont
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1996/
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This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules. Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.
The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation. Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event. The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files. It generates
the final core-[lang].xml file.
As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.
Review: https://reviewboard.asterisk.org/r/1967/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 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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While working with ast_parse_arg() to perform a validity check, a segfault
occurred. The segfault occurred due to passing a NULL pointer to
ast_sockaddr_parse() from ast_parse_arg(). According to the documentation in
config.h, "result pointer to the result. NULL is valid here, and can be used to
perform only the validity checks."
This patch fixes the segfault by checking for a NULL pointer. This patch also
adds documentation to netsock2.h about why it is necessary to check for a NULL
pointer.
(Closes issue ASTERISK-20006)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-20006-netsock-null-ptr.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1990/
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Channel drivers that allow native bridging need to handle
AST_CONTROL_PVT_CAUSE_CODE frames and previously did not handle them
properly, usually breaking out of the native bridge. This change
corrects that behavior and exposes the available cause code information
to the dialplan while native bridges are in place. This required
exposing the HANGUPCAUSE hash setter outside of channel.c, so
additional documentation has been added.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369061 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
Add support-level indications to many more source files.
Since we now have tools that scan through the source tree looking for files
with specific support levels, we need to ensure that every file that is
a component of a 'core' or 'extended' module (or the main Asterisk binary)
is explicitly marked with its support level. This patch adds support-level
indications to many more source files in tree, but avoids adding them to
third-party libraries that are included in the tree and to source files
that don't end up involved in Asterisk itself.
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r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
Add a script to enable finding source files without support-levels defined.
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Continuing with the Who Hung Up? project for Asterisk 11, this adds
support to IAX2 for the HANGUPCAUSE hash.
Additionally, this breaks out some functionality in frame.c for getting
information about frame types and subclasses.
Review: https://reviewboard.asterisk.org/r/1941/
(issue SWP-4222)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Make non-normal dialplan execution routines be able to run on a hung up
channel. This is preparation work for hangup handler routines.
* Fixed ability to support relative non-normal dialplan execution
routines. (i.e., The context and exten are optional for the specified
dialplan location.) Predial routines are the only non-normal routines that
it makes sense to optionally omit the context and exten. Setting a hangup
handler also needs this ability.
* Fix Return application being able to restore a dialplan location
exactly. Channels without a PBX may not have context or exten set.
* Fixes non-normal execution routines like connected line interception and
predial leaving the dialplan execution stack unbalanced. Errors like
missing Return statements, popping too many stack frames using StackPop,
or an application returning non-zero could leave the dialplan stack
unbalanced.
* Fixed the AGI gosub application so it cleans up the dialplan execution
stack and handles the autoloop priority increments correctly.
* Eliminated the need for the gosub_virtual_context return location.
Review: https://reviewboard.asterisk.org/r/1984/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The Hangup application used to just return -1 to cause normal dialplan
execution to hangup a channel. For the non-normal execution routines like
predial and connected-line interception routines, the hangup request would
exit the routine early but otherwise be ignored.
* Made the Hangup application not allow setting a cause code of zero. A
zero cause code is not defined.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added documentation describing what flags and arguments to pass to
aco_option_register for default option types. Also changed the ACL
handler to use the flags parameter to differentiate between "permit"
and "deny" instead of adding an additional vararg parameter.
Review: https://reviewboard.asterisk.org/r/1969/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
* Restructure ast_do_masquerade() to not hold channel locks while it calls
ast_indicate().
* Simplify many calls to ast_do_masquerade() since it will never return a
failure now. If it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is generate a
warning message about strange things may happen and press on.
* Fixed the call to ast_bridged_channel() in ast_do_masquerade(). This
change fixes half of the deadlock reported in ASTERISK-19801 between
masquerades and chan_iax.
(closes issue ASTERISK-19537)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1915/
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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/
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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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