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This framework adds a way to register the various options in a config
file with Asterisk and to handle loading and reloading of that config
in a consistent and atomic manner.
Review: https://reviewboard.asterisk.org/r/1873/
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* Fixes findings: 0-2,5,7-15,24-26,28-31
(issue ASTERISK-19648)
Reported by: Matt Jordan
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* Fix only issue pointed out by deprecated_REVERSE_INULL.txt for
app_meetme.c in find_user().
* Change use of %i to %d in sscanf() in find_user(). The use of %i gives
unexpected parsing because it can accept hex, octal, and decimal integer
formats.
* Changed other uses of %i in app_meetme() to use %d for consistency.
(issue ASTERISK-19648)
Reported by: Matt Jordan
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The Dial and Queue I option is intended to block connected line updates
and redirecting updates. However, it is a feature that when a call is
locally redirected, the I option is disabled if the redirected call runs
as a local channel so the administrator can have an opportunity to setup
new connected line information. Unfortunately, the Dial and Queue I
option is disabled for *all* forked calls if one of those calls is
redirected.
* Make the Dial and Queue I option apply to each outgoing call leg
independently. Now if one outgoing call leg is locally redirected, the
other outgoing calls are not affected.
* Made Dial not pass any redirecting updates when forking calls.
Redirecting updates do not make sense for this scenario.
* Made Queue not pass any redirecting updates when using the ringall
strategy. Redirecting updates do not make sense for this scenario.
* Fixed deadlock potential with chan_local when Dial and Queue send
redirecting updates for a local redirect.
* Converted the Queue stillgoing flag to a boolean bitfield.
(closes issue ASTERISK-19511)
Reported by: rmudgett
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1920/
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A patch introduced in r354938 made it so that ConfBridge would not attempt to
play sound files if those files did not exist. Unfortunately, ConfBridge uses
the same underlying function, play_sound_helper, to playback both sound files
and numbers to callers. When a number is being played back, the name of the
sound file is expected to be NULL. This NULL value was passed into a function
that tested for the existance of a sound file and is not tolerant to NULL
file names, causing a crash.
This patch fixes the behavior, such that if a sound file does not exist we
do not attempt to play it, but we only attempt that check if the a sound file
was specified in the first place. If a sound file was not specified, we use
the 'play number' logic in the helper function.
(closes issue ASTERISK-19899)
Reported by: Florian Gilcher
Tested by: Florian Gilcher
patches:
asterisk-19899.diff uploaded by mjordan (license 6283)
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Adds a number of methods for controlling the setting of 'ringinuse'
which is basically the same concept as the old ignorebusy setting,
only now the per member setting always controls whether or not the
member is actually ringed while in use. A CLI command and a manager
action have been added to change a given queue member's ringinuse
option while Asterisk is running and the an argument has been added
for adding members with deliberately set ringinuse in queues.conf
Some effort has been made to ensure compatability with dialplans and
databases still referring to 'ignorebusy'.
(issue ASTERISK-19536)
reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1919/
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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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Like the new predial feature for Dial. This adds the same b/B options to
FollowMe.
Review: https://reviewboard.asterisk.org/r/1910/
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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)
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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/
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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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(issue ASTERISK-19658)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1905/
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(issue ASTERISK-19652)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1909/
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If you hit the wrong DTMF digit trying to accept/decline a FollowMe call,
you had to wait for the prompt to repeat to try again.
* Make FollowMe compare the last DTMF digits received to the
accept/decline matching strings.
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The FollowMe caller call leg is usually answered and listening to MOH.
The caller could put the call on hold while FollowMe is looking for a
winner. The winning outgoing call is now immediately placed on hold if
the caller has put the call on hold before the winning call was selected.
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mallocing it.
Why this tiny struct was malloced instead of the 28k struct in the last
change is beyond me. Just doing my part to help stamp out sillyness.
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Sending the 'I' command from an external process will cause the current playlist
to be cleared, including stopping any audio file that is currently playing. This
is useful when you want to interrupt audio playback only when specific DTMF is
entered by the caller.
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Helping to stamp out stack abuse.
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* Fix FollowMe leaving recorded caller name file on error paths in
app_exec().
* Use correct buffer dimension define in struct fm_args.namerecloc[].
This fixes unexpected namerecloc filename length restriction.
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* Made use MAX_YN_STRING define to make all accept/decline DTMF buffers
the same size. Just using 20 isn't good enough when someone didn't get
the memo.
* Fix stupid use of a global variable in FollowMe. (ynlongest)
* Fix bit field declarations in FollowMe.
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Those channels are opaque now...
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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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Review: https://reviewboard.asterisk.org/r/1896/
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Correct the code in app_confbridge to play the conf-placeintoconf message to
the marked user entering the bridge instead of to the conference while the
marked user hears silence.
(closes issue ASTERISK-19641)
Reported-by: Mark A Walters
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The configuration option to specify a custom sound_leader_has_left file for a
conference bridge was not being parsed. This patch fixes it so that a custom
sound file will now be used.
(closes issue ASTERISK-19771)
Reported by: Pawel Kuzak
Tested by: Pawel Kuzak, Michael L. Young
Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak (license 6380)
Review: https://reviewboard.asterisk.org/r/1884/
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The warnings were about argv[0] being used uninitialized, which is correct.
Just remove setting username to this value, since username is set again before
it actually gets used.
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Thanks to Mark Murawski for the initial patch and feature definition.
(closes issue ASTERISK-19548)
Reported by: Mark Murawski
Review: https://reviewboard.asterisk.org/r/1878/
Review: https://reviewboard.asterisk.org/r/1229/
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* Delay duplicating a string on the stack in pickup_exec().
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Thanks Tilghman!
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For some reason, features.c has it's own definition. Should propably be fixed too.
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* Simplify some code in app_dial and app_queue by calling
ast_app_exec_macro() and ast_app_exec_sub().
* Fix minor locking issue in app_dial for post-answer macro/gosub
MACRO/GOSUB_RESULT=GOTO: handling.
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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.
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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.
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The Speech API apps return -1 on failure, which will hang up the channel. This
may not be desirable behavior for some, but it isn't something that can be
changed without breaking people's dialplans or writing an option to all of the
Speech apps that does what TryExec already does. This patch documents the
hangup behavior of the apps, and suggests TryExec as the solution.
(closes issue AST-813)
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Review: https://reviewboard.asterisk.org/r/1732/
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If the first command sent from an ExternalIVR client is an 'S' command, we were
blindly removing the first element from the play list and deferencing it, even
if it was NULL. This corrects that and also locks appropriately in one place.
(issue ASTERISK-17889)
Reported by: Chris Maciejewski
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* 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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Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 362496 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Patch by: junky
Review: https://reviewboard.asterisk.org/r/1743/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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