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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
........
Merged revisions 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 365475 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365477 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)
........
Merged revisions 365398 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 365399 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@365400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/
........
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
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
* 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
* 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
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
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)
........
Merged revisions 362815 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 362816 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
........
Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
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
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
........
Merged revisions 362201 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 362202 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
........
Merged revisions 362082 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 362084 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
........
Merged revisions 361558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 361560 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
........
Merged revisions 360033 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
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
* 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
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
........
Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........
Merged revisions 359656 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
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
When connected line support was added, the wait_for_answer() variable
single changed its meaning slightly. Unfortunately, the places where
single was used did not necessarily get updated to reflect that change.
Also audio/video frames were sent to all forked calls when the endpoints
were never made compatible.
* Don't pass audio/video media frames when the channels have not been made
compatible.
* Added handling of AST_CONTROL_SRCCHANGE to app_dial.c.
* Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also
pass a requested MOH class.
(closes issue ASTERISK-16901)
Reported by: Chris Gentle
(closes issue ASTERISK-17541)
Reported by: clint
Review: https://reviewboard.asterisk.org/r/1805/
........
Merged revisions 359344 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 359355 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359357 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/
........
Merged revisions 358810 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 358811 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This should (and does in my testing) act just like the Dial option of the same name.
This allows a queue member to be transfered to the next priority (no args), or to
a context/extension/priority similar to goto (with args context^extension^priority)
when a caller hangs up on them.
(closes issue ASTERISK-19283)
Reported by: To
Patches:
queue_f-v3.diff uploaded by To (license 6347)
Review: https://reviewboard.asterisk.org/r/1785/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When the stack frame is allocated, we save the address to which we should
return, when the Gosub returns. However, if we just want to restore the
priority, then we need to subtract 1 before setting it. Otherwise, when
a Gosub goes to a nonexistent address, it will skip a priority in the
dialplan. This is because when we return from an application, the PBX
increments the priority for us.
........
Merged revisions 357416 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 357421 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated. This also adds
deprecation warnings for those features when used and in documentation.
Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers. However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL. In that case, an invalid free would be attempted,
which could crash app_voicemail. As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers. This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
........
Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 356798 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This simply fixes the compilation issue introduced in r354429 by
re-adding the 'quote' variable.
(closes issue ASTERISK-19337)
Reported by: John Taylor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Continue channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these variables, but
the purpose for now is to hide the implementation and keep people from
adding code that directly accesses the channel structure. Semantic
changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
MixMonitor receives a new option i(channel_variable) which stores the unique id at said
variable. StopMixMonitor now accepts ID as an optional argument, which if included will
make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI
commands and AMI actions have been ammended to work with the IDs as well. In addition,
monitors across a channel can now be listed be listed via CLI command "mixmonitor list
<channel>" which will display all of the mixmonitors active on that channel along with
the files they each have open. Created by Sergio González Martín.
(closes issue ASTERISK-19096)
Reported by: Sergio González Martín
Review: https://reviewboard.asterisk.org/r/1643/
Review: https://reviewboard.asterisk.org/r/1682/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, realtime queues could be loaded without defining the queue member
table. This allowed for queue members to be dynamic, while the realtime
queue definitions could exist in some backing storage. Revision 342223 broke
this when it changed the return value for realtime_multientry to return NULL
when no results are returned. Previously, an empty ast_config object was
expected.
(closes issue ASTERISK-19170)
Reported by: Rene Mendoza
Tested by: Rene Mendoza
Patches:
rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283)
........
Merged revisions 350552 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 350553 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch addresses two issues in ConfBridge and the channel bridge layer:
1. It fixes a race condition wherein the bridge channel could be hung up
2. It removes the deadlock avoidance from the bridging layer and makes the
bridge_pvt an ao2 ref counted object
Patch by David Vossel (mjordan was merely the commit monkey)
(issue ASTERISK-18988)
(closes issue ASTERISK-18885)
Reported by: Dmitry Melekhov
Tested by: Matt Jordan
Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628)
(closes issue ASTERISK-19100)
Reported by: Matt Jordan
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1654/
........
Merged revisions 350550 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.
This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.
The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.
The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).
Review: https://reviewboard.asterisk.org/r/1655/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When looking up a ConfBridge profile, the config parser would, if it
found a channel datastore on the channel requesting the bridge profile,
unlock the channel mutex twice. Since that's a little aggressive,
it now only unlocks it once.
(closes issue ASTERISK-19042)
Reported by: Matt Jordan
Tested by: Matt Jordan
Patches:
19042 uploaded by David Vossel (license 5628)
........
Merged revisions 349619 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch allows the imapserver, imapport, and imapflags settings to be
overridden for any voicemail user. It also documents the settings in
the sample voicemail.conf file, and updates the voicemail schema to
allow storage of those columns.
(closes issue ASTERISK-16489)
Reporter: Hubert Mickael
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1614/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
................
r349045 | seanbright | 2011-12-23 12:32:33 -0500 (Fri, 23 Dec 2011) | 25 lines
Merged revisions 349044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) | 18 lines
In ChanSpy, don't create audiohooks that will never be used.
When ChanSpy is initialized it creates and attaches 3 audiohooks:
1) Read audio off of the channel that we are spying on
2) Write audio to the channel that we are spying on
3) Write audio to the channel that is bridged to the channel that we are
spying on.
The first is always necessary, but the others are used only when specific
options are passed to the ChanSpy application (B, d, w, and W to be specific).
When those flags are not passed, neither of those audiohooks are ever sent
frames, but we still try to process the hooks for each voice frame that we
recieve on the channel.
So in short - only create and attach audiohooks that we actually need.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to check the availability of the caller's name, app_voicemail will check for an
audio file in <astspooldir>/recordings/callerids/
This change sets a precedent for where to put recordings of names. Currently the idea is
that recordings here could also be used for applications like confbridge and meetme to
find recorded names in this folder from callerid (when another recording isn't available)
(closes issue ASTERISK-18565)
Reporter: Russell Brown
Patches:
r uploaded by Russel Brown (license 6182)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ParkAndAnnounce tried to pass the CallerID to the announcing channel but
the ID was wiped out by the channel masquerade done when parking the call.
* Save the CallerID before parking the channel to pass it to the
announcing channel.
* Fixed a minor memory leak in ParkAndAnnounce.
* Updated some ParkAndAnnounce log messages.
........
Merged revisions 348310 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 348311 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, app_originate could not originate a call into a non-8kHz conference
bridge as the formats for non-8kHz slin codecs were not applied to the created
channel. This patch adds all of the formats by default, such that if a created
channel has a codec that supports a higher sampling rate, a translation path
can be built between it and other channels.
........
Merged revisions 348265 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The function QUEUE_MEMBER has two required parameters (queuename, option). It
was only checking for the presence of queuename. The patch checks for the
existence of the option parameter and provides better error logging when
invalid values are provided for the option parameter as well.
........
Merged revisions 348211 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Provided a more descriptive error message when a value supplied for the parameter
type is not one of the acceptable values.
(closes issue ASTERISK-18717)
Reported by: Paul Belanger
Patches:
__20111103-better-confbridge_info-error-msg.txt (License #4999)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The addition of the Connected Line support changed how CallerID is passed
to outgoing calls. The FollowMe application was not updated to pass
CallerID to the outgoing calls.
* Fix FollowMe CallerID on outgoing calls.
* Restructured findmeexec() to fix several memory leaks and eliminate some
duplicated code.
* Made check the return value of create_followme_number(). Putting a NULL
into the numbers list is bad if create_followme_number() fails.
* Fixed a couple uses of ast_strdupa() inside loops.
* The changes to bridge_builtin_features.c fix a similar CallerID issue
with the bridging API attended and blind transfers. (Not used at this
time.)
(closes issue ASTERISK-17557)
Reported by: hamlet505a
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1612/
........
Merged revisions 348101 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 348102 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348103 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These commands work much like the dialplan applications that would otherwise invoke them.
A nice benefit of these is that they can be invoked on a call remotely and at any time
during a call. They work much like the Monitor and StopMonitor ami commands.
(closes issue ASTERISK-17726)
Reported by: Sergio González Martín
Patches:
mixmonitor_actions.diff uploaded by Sergio González Martín (license 5644)
Review: https://reviewboard.asterisk.org/r/1193/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a caller sends DTMF while the SayUnixTime application is saying the time, The call
would jump to the next extension much like it does during Background(). This patch adds
option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch
allows arguments to sayunixtime to not be used as empty strings in the case of something
like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern).
(closes issue ASTERISK-16675)
Reported by: jlpedrosa
Patches:
patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959)
Review: https://reviewboard.asterisk.org/r/956/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when
attempting to set penalty on an interface without specifying a queue in the queue set
penalty CLI command. In addition, no attempt would be made whatsoever to perform the
penalty setting on all the queues in the core list with either the cli command or the
non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to
document and rename some functions required by this command to better represent what
they actually do. Oh yeah, and the use of this command without specifying a specific
queue actually works now.
Review: https://reviewboard.asterisk.org/r/1609/
........
Merged revisions 347656 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Meetme would attempt to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the channel variable set
for those variables in spite of those database entries being NULL or even lacking
a column to represent them.
(closes issue ASTERISK-18873)
Reported by: Byron Clark
Patches:
ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157)
........
Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 347383 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count. The admin action will globally mute /
unmute all non-admin participants on a converence, while the participant
count simply exposes the existing participant count function to the
conference bridge menu.
This also adds configuration options to change the sound played when the
conference is globally muted / unmuted, as well as the necessary config
hooks to place these functions in the DTMF menus.
(closes issue ASTERISK-18204)
Reported by: Kevin Reeves
Tested by: Matt Jordan
Patches:
app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
Review: https://reviewboard.asterisk.org/r/1518/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It was strange that the AgentCalled AMI event would get most of its
information from the incoming channel but then get the CallerID
information from the outgoing channel. Before connected line support was
added, this information was always the same at this point.
(closes issue ASTERISK-18152)
Reported by: Thomas Farnham
Tested by: rmudgett
........
Merged revisions 344536 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 344537 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The MeetMe application documentation has some comments about usage of DAHDI,
and they were a bit outdated relative to modern DAHDI releases. This patch
changes the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since that module
does not exist in current DAHDI releases.
........
Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 342991 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Despite an ominous sounding comment stating that membercount was for "logged
in" members only and thus we couldn't use ao2_container_count(), I could not
find a single place in the code where that seemed to be accurate. The only time
we decremented membercount was when we were marking something dead or actually
removing it. The only places we incremented it were either after ao2_link(), or
trying to correct for having set it to 0 during a reload. In every case where
we were correcting the value, it seemed that we were trying to make the count
actually match what ao2_container_count() would return. The only place I could
find where we made a determination about something being "logged in" or not, we
didn't trust the membercount, but instead looked at devicestate, paused, etc.
This patch removes membercount, replaces its use with ao2_container_count, and
manually adds the results of ao2_container_count to a "membercount" field for
ast_data queue query results. This patch also would fix AST-676, but as it is
slightly riskier than the previously committed fix, the two commits have been
made separately.
Reivew: https://reviewboard.asterisk.org/r/1541/
........
Merged revisions 342383 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 342384 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
To use the new OBJ_KEY flag, the container hash and compare callback
functions must be updated to support OBJ_KEY. Otherwise, bad things
happen.
(issue ASTERISK-14769)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
........
r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) | 15 lines
Add option to check state when state is unknown
r341486 reverts r325483 this is a rework of the patch.
optimize to minimize load.
add option check_state_unknown to control whether a member with unknown
device state is checked there is a small % chance that calls will be sent
to the member when they on a call.
app_queue will see a device with unknown state as available and does not
try verify the state without this option enabled.
Review: https://reviewboard.asterisk.org/r/1535/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/10
........
r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct 2011) | 18 lines
Fix a performance regression introduced in r325483.
The regression was caused by a call to ast_parse_device_state() in app_queue's
ring_entry() function. The ast_parse_device_state() function eventually calls
ast_channel_get_full() with a channel name prefix which causes it to walk the
channel list causing massive lock contention and slow downs.
This patch fixes the regression by removing the call to
ast_parase_device_state() which should be unnecessary. Queue member device
state should be maintained by device state events. Some users have seen
instances where busy agents were called when they shouldn't have, which is the
reason the call to ast_parse_device_state() was added. That change appears to
have resolved that issue but also causes this performance regression. There may
still be issues with queue member status, and if so, alternative methods should
be investigated to resolve them.
AST-695
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Jonathan Rose
Richard Mudgett
Sean Bright
Matthew Jordan
Kinsey Moore
Michael L. Young
Russell Bryant
Olle Johansson
Terry Wilson
Walter Doekes
Mark Michelson
Mark Murawki
Tilghman Lesher
Joshua Colp
Paul Belanger
Jason Parker
TransNexus OSP Development
Leif Madsen
Kevin P. Fleming
Gregory Nietsky
Matthew Nicholson