The ability to set "echocan_mode" and "buffers" through the dialplan was added
to chan_dahdi some time ago. This patch adds some documentation to
func_channel.
(Closes issue ASTERISK-19911)
Reported by: Dale Noll
Tested by: Michael L. Young
Patches:
asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1949/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@368092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
(issue ASTERISK-19665)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1922/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366944 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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366880 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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
As it turned out, this wasn't a huge deal. We were calling
ast_app_parse_options() for a set of options of which none
took arguments. The proper thing to do for this case is to
pass NULL for the "args" parameter here. We were instead passing
a seemingly-randomly chosen char * from the function. While this
would never get written to, you can rest assured things would
have gotten bad had new options (which took arguments) been added
to func_volume.
(closes issue ASTERISK-19656)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@364899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In a variety of locations in both reading and writing a file, the result
from the C library function ftello is used as input to other functions. For
the parameters and functions in question, a negative value is invalid input.
This patch checks the return value from the ftello function to determine if
we were able to determine the current position in the file stream and, if not,
fail gracefully.
(issue ASTERISK-19655)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1863/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the global_curl_info data structure could not be allocated, the
datastore associated with the operation would be free'd, but the function
would not return. This would later dereference the datastore, almost
certainly causing Asterisk to crash. With this patch, if the data
structure is not allocated the method will return an error code, and
not attempt any further operation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361753 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the SHARED function modifies a variable, it removes it from its list of
variables and reinserts the new value at the head of the list of variables.
Doing this inside a standard list traversal can be dangerous, as the
standard list traversal does not account for the list being changed. While
the code in question should not cause a use after free violation due to its
breaking out of the loop after freeing the variable, it could lead to a
maintenance issue if the loop was modified. This also fixes a violation
reported by a static analysis tool, which also makes this code easier to
maintain in the future.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361657 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change restores functionality that was present in 1.4, when AEL macros
were implemented with the Macro dialplan application. Macros are fraught with
functionality issues, because they consume a large portion of the underlying
application stack. This limits the ability of AEL users to call many layers
of subroutines, an issue which Gosub does not have (originally tested to
100,000 levels deep). Therefore, starting in 1.6.0, AEL macros were
implemented with Gosub.
However, there were some implicit behaviors of Macro, which were not replicated
at the same time as with the transition to Gosub, one of which is documented in
the related issue. In particular, the "h" extension is designed to execute not
in the Macro context, but in the topmost calling context. Due to legacy issues
with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks
in all calling contexts, bubbling up from the deepest level until it finds an
"h" extension.
Since AEL hides the complexity of the underlying dialplan logic from the AEL
programmer, it's reasonable to assume that this behavior should not change in
the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break
working AEL configurations in the transition to Asterisk 1.8 LTS. This fix
is the result, which implements a search for the "h" extension in all calling
Gosub contexts.
Fixes ASTERISK-19336
Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher
(with slight modifications for 1.8)
Tested by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1776/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
CDRs cannot be modified after a bridge is torn down, (e.g. after
Dial() returns) even though the CDR() function may be called. Since
modifying the CDR code to change this behavior could very easily
break all kinds of things, this patch just documents this limitation.
(closes issues ASTERISK-16923)
Review: https://reviewboard.asterisk.org/r/1720/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
For some reason this function was completely undocumented in 1.8. I copied the
10 docs over to 1.8 and removed references to an enumerator that was added in
the Asterisk 10 version of func_curl. That was the only change I noted.
(closes issue ASTERISK-19186)
Reported by: Olivier Krief
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note: Noone calls ast_app_dtget() with the timeout parameter of zero so
the bad code normally will never get executed.
* Fix unnecessary floating point division in func_timeout.c
timeout_write() when all other values are integers.
(closes issue ASTERISK-16817)
Reported by: Dmitry Andrianov
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The time passed by the LOCK function to an internal function was relative
time when the function expected absolute time.
* Don't use C++ keywords in get_lock().
(closes issue ASTERISK-16868)
Reported by: Andrey Solovyev
Patches:
20101102__issue18207.diff.txt (license #5003) patch uploaded by Andrey Solovyev (modified)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@350311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to
be called by different threads for the same channel. The channel driver
thread and the PBX thread running dialplan.
* Add lock protection around CDR API calls that access an ast_channel
pointer.
(closes issue ASTERISK-18836)
Reported by: gpluser
Review: https://reviewboard.asterisk.org/r/1628/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dummy channels created by ast_dummy_channel_alloc() should be destoyed by
ast_channel_unref(). Using ast_channel_release() needlessly grabs the
channel container lock and can cause a deadlock as a result.
* Analyzed use of ast_dummy_channel_alloc() and made use
ast_channel_unref() when done with the dummy channel. (Primary reason for
the reported deadlock.)
* Made app_dial.c:dial_exec_full() not call ast_call() holding any channel
locks. Chan_local could not perform deadlock avoidance correctly.
(Potential deadlock exposed by this issue. Secondary reason for the
reported deadlock since the held lock was part of the deadlock chain.)
* Fixed some uses of ast_dummy_channel_alloc() not checking the returned
channel pointer for failure.
* Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected
by testing the bogus_chan value.
* Fixed needlessly clearing a 1024 char auto array when setting the first
char to zero is enough in manager.c:action_getvar().
(closes issue ASTERISK-18613)
Reported by: Thomas Arimont
Patches:
jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Thomas Arimont
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The return value of popen() was not checked for failure to open.
(closes issue ASTERISK-18109)
JIRA SWP-3633
Reported by: Michael Myles
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Starting with Asterisk v1.8, the DAHDI channel name format was changed for
ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
There were several reasons that the channel name had to change.
1) Call completion requires a device state for ISDN phones. The generic
device state uses the channel name.
2) Calls do not necessarily have B channels. Calls placed on hold by an
ISDN phone do not have B channels.
3) The B channel a call initially requests may not be the B channel the
call ultimately uses. Changes to the internal implementation of the
Asterisk master channel list caused deadlock problems for chan_dahdi if it
needed to change the channel name. Chan_dahdi no longer changes the
channel name.
4) DTMF attended transfers now work with ISDN phones because the channel
name is "dialable" like the chan_sip channel names.
For various reasons, some people need to know which B channel a DAHDI call
is using.
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
in use by the channel. Use CHANNEL(no_media_path) to determine if the
channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an
Asterisk channel so AMI applications can passively determine the B channel
currently in use. Calls with "no-media" as the DAHDIChannel do not have
an associated B channel. No-media calls are either on hold or
call-waiting.
(closes issue #17683)
Reported by: mrwho
Tested by: rmudgett
(closes issue #18603)
Reported by: arjankroon
Patches:
issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: stever28, rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added XML documentation for CHANNEL(keypad_digits) and
CHANNEL(no_media_path).
* Tweaked XML documentation for CHANNEL(reversecharge).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309170 65c4cc65-6c06-0410-ace0-fbb531ad65f3
