Commit Graph

4127 Commits

Author SHA1 Message Date
Richard Mudgett 1b0428ac7d * Fix accept/decline DTMF buffer overwrite in FollowMe.
* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-08 18:16:04 +00:00
Matthew Jordan 11faa15d11 Fix channel opaquification slip-up in r365477
Those channels are opaque now...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07 18:58:40 +00:00
Matthew Jordan 9e7de73fee Support VoiceMail d() option when extension does not exist in channel's context
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-07 18:42:48 +00:00
Kinsey Moore 781f4657b9 Fix many issues from the NULL_RETURNS Coverity report
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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Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-04 22:17:38 +00:00
Sean Bright 474612d7f7 Add IPv6 support to ExternalIVR.
Review: https://reviewboard.asterisk.org/r/1896/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-03 14:47:58 +00:00
Kinsey Moore a965f18695 Play conf-placeintoconf message to the correct channel
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-05-01 19:10:48 +00:00
Michael L. Young 2cbcbc7f6b Fix configuring custom sound_leader_has_left in confbridge.conf
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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Merged revisions 364536 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-29 02:23:22 +00:00
Russell Bryant a498ef2aa0 app_minivm: Fix a couple compiler warnings.
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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28 01:10:35 +00:00
Richard Mudgett e8a6e0ef0e PreDial - Ability to run dialplan on callee and caller channels before Dial.
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/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28 00:31:47 +00:00
Richard Mudgett 238640dc1b Update Pickup application documentation. (With feeling this time.)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-26 21:11:25 +00:00
Olle Johansson e5c20ccb76 Code formatting fixes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-26 13:59:11 +00:00
Richard Mudgett 9d655bd0e8 Update Pickup application documentation. (Even better)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-26 03:12:44 +00:00
Richard Mudgett e736a4fed3 * Put more information in pickup_exec() LOG_NOTICE.
* Delay duplicating a string on the stack in pickup_exec().


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-26 01:29:09 +00:00
Richard Mudgett 0986873128 Update Pickup application documentation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363790 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25 23:00:26 +00:00
Olle Johansson 04ddb5714f Add documentation
Thanks Tilghman!


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25 13:57:01 +00:00
Olle Johansson f102aecf12 Formatting changes only
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25 11:18:14 +00:00
Olle Johansson a8e755700e Use the DEFINED value for musicclass length.
For some reason, features.c has it's own definition. Should propably be fixed too.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-25 10:49:13 +00:00
Richard Mudgett f663924517 Make app_dial and app_queue use new macro and gosub calls.
* 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.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@363269 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-23 17:05:55 +00:00
Richard Mudgett c870dad57e Update app_dial M and U option GOTO return value documentation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362999 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-21 01:46:34 +00:00
Richard Mudgett 3a874139d4 Fix connected-line/redirecting interception gosubs executing more than intended.
* Redo ast_app_run_sub()/ast_app_exec_sub() to use a known return point so
execution will stop after the routine returns there.
(s@gosub_virtual_context:1)

* Create ast_app_exec_macro() and ast_app_exec_sub() to run the macro and
gosub application respectively with the parameter string already created.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20 23:29:56 +00:00
Richard Mudgett 01194c5811 Use ast_channel_lock_both() where it was inlined before.
The CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the channel
lock was originally obtained is overkill where ast_channel_lock_both() was
inlined.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20 16:23:01 +00:00
Terry Wilson 34d670f786 Document Speech* apps hangup on failure and suggest TryExec
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-20 14:50:42 +00:00
Terry Wilson 6d6bacd5cb Convert some strncpys to ast_copy_string
Review: https://reviewboard.asterisk.org/r/1732/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-19 19:05:17 +00:00
Sean Bright ba93541ced Prevent a crash in ExternalIVR when the 'S' command is sent first.
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-19 16:10:04 +00:00
Matthew Jordan f78290068a Fix a variety of potential buffer overflows
* 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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-19 02:40:55 +00:00
Walter Doekes fc63e07135 Avoid cppcheck warnings; removing unused vars and a bit of cleanup.
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
2012-04-17 18:57:40 +00:00
Matthew Jordan ebcccf8997 Fix handling of negative return code when storing voicemails in ODBC storage
When storing a voicemail message using an ODBC connection to a database, the
voicemail message is first stored on disk.  The sound file associated with
the message is read into memory before being transmitted to the database.
When this occurs, a failure in the C library's lseek function would cause a
negative value to be passed to the mmap as the size of the memory map to
create.  This would almost certainly cause the creation of the memory map to
fail, resulting in the message being lost.

(issue ASTERISK-19655)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1863
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-16 21:42:12 +00:00
Jonathan Rose ba0f044bde Make ForkCDR e option not set end time of the newly forked CDR log
Prior to this patch, ForkCDR's e option would immediately set the end time of the forked
CDR to that of the CDR that is being terminated. This resulted in the new CDR's end time
being roughly the same as it's beginning time (which is in turn roughly the same as the
original's end time).

(closes issue ASTERISK-19164)
Reported by: Steve Davies
Patches:
	cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-13 16:12:17 +00:00
Jonathan Rose c0b9fe8530 Send relative path named recordings to the meetme directory instead of sounds
Prior to this patch, no effort was made to parse the path name to determine a proper
destination for recordings of MeetMe's r option. This fixes that.

Review: https://reviewboard.asterisk.org/r/1846/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-13 15:38:08 +00:00
Jonathan Rose 683eacb59a Change default value of 'ignorebusy' on Queue members so that behavior is more like 1.8
Prior to this patch, in order to restore that behavior, a function would have
to be used on the QueueMember to make the ringinuse option do anything, which
is pretty unreasonable.

(closes issue ASTERISK-19536)
reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1860/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-11 17:20:08 +00:00
Matthew Jordan aa21d4fc6b Fix memory leak when using MeetMeAdmin 'e' option with user specified
A memory leak/reference counting leak occurs if the MeetMeAdmin 'e' command
(eject last user that joined) is used in conjunction with a specified user.
Regardless of the command being executed, if a user is specified for the
command, MeetMeAdmin will look up that user.  Because the 'e' option kicks
the last user that joined, as opposed to the one specified, the reference to
the user specified by the command would be leaked when the user variable
was assigned to the last user that joined.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 20:32:52 +00:00
Kinsey Moore a485f44022 Add missing newlines to CLI logging
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 18:19:03 +00:00
Russell Bryant b2f7b0c649 Remove a few more files related to chan_usbradio and app_rpt.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 15:50:18 +00:00
Kinsey Moore 51f0e5c53d Remove unnecessary error message in app_dial.c
The error message for failure to stop autoservice after a gosub or macro call
during a dial was removed for macro while Asterisk 1.4 was still being actively
developed. The corresponding gosub error message was never removed.

(closes issue ASTERISK-19551)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-06 13:32:34 +00:00
Jonathan Rose fc45af331b Fix MusicOnHold in MeetMe so that it always uses the class if it's been defined
There were a few instances of restarting music on hold in meetme that would cause
Asterisk to revert to the default class of music on hold for no adequate reason.

Review: https://reviewboard.asterisk.org/r/1844/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-05 17:22:30 +00:00
Jonathan Rose e96a59acfd Replace GNU old-style field designator extensions to fix clang warnings
(issue ASTERISK-19540)
Reported by: Makoto Dei
Patches:
	clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
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Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8

Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8
(closes issue ASTERISK-19540)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04 18:08:28 +00:00
Jonathan Rose 97b2fa8de1 Make the MeetMeAdmin N command (mute all nonadmins) not mute admins
(Closes Issue ASTERISK-19335)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1843/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-04 13:51:45 +00:00
Kinsey Moore 93781fa161 Fix the display of documentation for Transfer
This came up while fixing documentation generation for many other cases where
the argument separator was not being displayed properly.  Now that it is
displayed properly, it shows up in the wrong place for Transfer since the '/'
is only required if Tech is present.

(related to issue ASTERISK-18168)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-03 20:14:01 +00:00
Jonathan Rose 655a8d4420 Introducing the log message unique call identifiers feature
Log messages will now display a call number that they are tied to (ordered for calls
based on when they started). This feature is made to be minimally invasive without
requiring changes to many of the existing log messages. These IDs  won't show up for
verbose messages on CLI (but they will in log files) This is currently in phase II
of production, see more about this feature on the wiki --
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging

Review: https://reviewboard.asterisk.org/r/1823/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 20:01:20 +00:00
Jonathan Rose d501c2ea2d undoing 360785 due to merging mistake
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 19:59:30 +00:00
Jonathan Rose bf994f0e04 Introducing the log message unique call identifiers feature
Log messages will now display a call number that they are tied to (ordered for calls
based on when they started). This feature is made to be minimally invasive without
requiring changes to many of the existing log messages. These IDs  won't show up for
verbose messages on CLI (but they will in log files) This is currently in phase II
of production, see more about this feature on the wiki --
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging

Review: https://reviewboard.asterisk.org/r/1823/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-29 19:54:35 +00:00
Terry Wilson dd9405db05 Fix setting CDR variables in the hangup extension
A previous CDR fix for setting CDR variables during a bridge via
custom dialplan features broke setting CDR variables in the
hangup extension. This patch fixes the issue.

Review: https://reviewboard.asterisk.org/r/1794/
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Merged revisions 358978 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 358989 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360724 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-28 19:39:24 +00:00
Russell Bryant 0ec73946fa app_page: Fix a memory leak on every Page().
dial_list is a dynamically allocated array that is allocated at the beginning
of Page() based on how many devices will be dialed.  This was never being freed.
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Merged revisions 360363 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 360364 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-24 03:11:43 +00:00
Russell Bryant 71a1541b0c app_jack: fix datastore memory leak in error handling path.
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Merged revisions 360360 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 360361 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-24 03:03:20 +00:00
Jonathan Rose c6979ff581 Adds F option to Bridge application
Similar to dial and queue F option.

(Closes issue ASTERISK-19282)
Reported by: To
Patches:
	bridge_f-v3.diff uploaded by To (license 6347)
Review: https://reviewboard.asterisk.org/r/1825/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-22 21:25:22 +00:00
Kinsey Moore 6ff8f14865 Prevent Echo() from relaying control, null, and modem frames
Echo()'s description states that it echoes audio, video, and DTMF except for #
while it actually echoes any frame that it receives other than DTMF #.  This
was causing frame storms in the test suite in some circumstances where Echo()
was attached to both ends of a pair of local channels and control frames
were being periodically generated.  Echo()'s behavior and description have
been modifed so that it only echoes media and non-# DTMF frames.
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Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 360034 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-20 20:42:34 +00:00
Jonathan Rose 0399daaa2e Prevent chanspy from binding to zombie channels
This patch addresses a bug with chanspy on local channels which roughly 50% of the time
would create a situation where chanspy can latch onto a zombie channel, keeping the zombie
alive forever and causing the channel doing the spying to never be able to hang up.

(closes issue ASTERISK-19493)
Reported by: lvl
Review: https://reviewboard.asterisk.org/r/1819/
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Merged revisions 359892 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359898 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-16 21:00:07 +00:00
Mark Michelson 827f2eae92 Revert the pre-dial addition.
The code may be just fine, but it had not received a "ship it!" on
review board yet.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-16 15:38:45 +00:00
Mark Murawki c65b41f57a Add options PreDial options 'b' and 'B' to app_dial
* Added 'b' and 'B' options to Dial.  These options will allow you to run
  last-minute dialplan on the caller and callee channels while the Dial
  application is executing, but before the call is started.  For example you
  can use the 'b' option to run dialplan on the callee channel to get the name
  of the newly created channel right away.

Review: https://reviewboard.asterisk.org/r/1229/

(closes issue: ASTERISK-19548)
Reported by: Mark Murawski
Tested by: Mark Murawski, Stefan Schmidt



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-15 18:58:25 +00:00
Matthew Jordan c61d49d5cc Fix remotely exploitable stack overrun in Milliwatt
Milliwatt is vulnerable to a remotely exploitable stack overrun when using
the 'o' option.  This occurs due to the milliwatt_generate function not
accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of
samples it can put in the output buffer.

This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET
when determining the maximum number of samples allowed.  Note that at no
point is remote code execution possible.  The data that is written into the
buffer is the pre-defined Milliwatt data, and not custom data.

(closes issue ASTERISK-19541)
Reported by: Russell Bryant
Tested by: Matt Jordan
Patches:
  milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283)
  Note that this patch was written by Russell, even though Matt uploaded it
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Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Merged revisions 359656 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 359694 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359704 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-15 18:55:54 +00:00