When Milliwatt() was changed internally to use Playtones() so that the proper
tone was used, it introduced a drop in gain in the output signal. So, use
the playtones API directly and specify a volume argument such that the output
matches the gain of the original Milliwatt() code.
(closes issue #15386)
Reported by: rue_mohr
Patches:
issue_15386.rev2.diff uploaded by russell (license 2)
Tested by: rue_mohr
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I know what some of you are thinking: "UGH! Mark, why are you using
ast_strdup and ast_free for the string when you can just use ast_strdupa
and let the memory free itself?! Have the bats been chewing on your brain
again?"
Based on past experiences, I don't like using ast_strdupa inside a loop.
It's a good way to potentially exhaust stack space. Also, since this only
happens when reloading queues, I don't think that heap allocations and
frees are going to be a huge problem.
(closes issue #15559)
Reported by: amorsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@208622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would incorrectly
indicate that the caller was still in the queue. With these changes,
the problem does not occur.
(closes issue #14631)
Reported by: latinsud
Patches:
queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
(with small modification from me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@205349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented
as seconds.
Also, the inequality was reversed. The warning, if triggered, was "Max silence should
be less than minmessage or you may get empty messages", which should have been logged
if max silence was greater than minmessage, but the check was for less than.
Also, conforming if statement to coding guidelines.
closes issue #15331)
Reported by: markd
Review: https://reviewboard.asterisk.org/r/293/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@203719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
StopMixMonitor only indicates to the MixMonitor thread to stop
writing to the file. It does not guarantee that the recording's
file handle is available to the dialplan immediately after execution.
This results in a race condition. To resolve this, the filestream
pointer is placed in a datastore on the channel. When StopMixMonitor
is called, the datastore is retrieved from the channel and the
filestream is closed immediately before returning to the dialplan.
Documentation indicating the use of StopMixMonitor to free files
has been updated as well.
(closes issue #15259)
Reported by: travisghansen
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/283/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@201423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@200991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize awat the Local
channel when using this option.
(issue #14829)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
command.
This change relates to r184980, which was a backport of the state interface
changes to app_queue from trunk. trunk and all of the 1.6.x branches are not
affected.
'queue add member' allows for specifying an interface to use for device state
when adding a queue member via CLI, but the validation code was not properly
updated to reflect this optional argument.
(closes issue #15198)
Reported by: loloski
Patches:
05272009_app_queue.diff uploaded by seanbright (license 71)
Tested by: loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@197024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
situations.
This involves setting a proper cdr disposition coresponding to the given
failure condition and ensuring the proper information is stored in the cdr
record.
(closes issue #13691)
Reported by: dferrer
Tested by: mnicholson
(closes issue #13637)
Reported by: atis
Tested by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows more concurrent extensions to be copied for a single voicemail,
without creating a possibility of upsetting existing users, where a dialplan
could run out of stack space where it had run fine before. Alternatively,
we could have allocated off the heap, but that is a larger change and would
have increased the chance for instability introduced by this change.
This is really solved starting in 1.6.0.11, as the use of an ast_str buffer
allows an unlimited number of extensions (up to available memory). We
additionally create a new warning message when the buffer length is exceeded,
permitting administrators to see an issue after the fact, whereas previously
the list was silently truncated.
(closes issue #14739)
Reported by: p_lindheimer
Patches:
20090417__bug14739.diff.txt uploaded by tilghman (license 14)
Tested by: p_lindheimer
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This fixes a case where a certain message could get played twice.
(closes issue #13155)
Reported by: greenfieldtech
Patches:
app_voicemail.c.multi-lang-patch uploaded by greenfieldtech (license 369)
Tested by: greenfieldtech
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Similar to seanbright's commit 191422, this moves some static buffers
to be defined outside of for loops since it is undefined if memory
will be re-used or if the stack will grow with each iteration of the
loop.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
According to Kevin, it is unspecified as to whether a variable defined inside
a block is allocated once by the compiler or for each pass through the block
(loops being the only interesting case), so just define these before we get
into our loop to be sure.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191422 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A user reported via #asterisk that with very long lists of members, a crash
occurs in ast_strdupa, so just use a single buffer and ast_copy_string instead
of stack allocating copys of each interface name.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@191041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Issue #14359 was fixed between the time that I posted the review of the backport
of the state interface change for 1.4. This merges the changes from that issue
back into 1.4.
(closes issue #14359)
Reported by: francesco_r
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(This is copied and pasted from the review request I made for this patch)
Asterisk has some odd behavior when queue weights are used. The current logic used when
potentially calling a queue member is:
If the member we are going to call is part of another queue and _that other queue has any
callers in it_ and has a higher weight than the queue we are calling from, then don't try
to contact that member. The issue here is what I have marked with underscores. If the
higher-weighted queue has any callers in it at all, then the queue member will be unreachable
from the lower-weighted queue. This has the potential to be really really bad if using a
queue strategy, such as leastrecent or fewestcalls, with the potential to call the same
member repeatedly.
The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works
well for this situation. With this set of changes, the logic used becomes:
If the member we are going to call is part of another queue, the other queue has a higher
weight than the queue we are calling from, and the higher weight queue has at least as many
callers as available members, then do not try to contact the queue member. If the higher
weighted queue has fewer callers than available members, then there is no reason to deny
the call to this member since the other queue can afford to spare a member.
Since the fix involved writing a generic function for determining the number of available
members in the queue, I also modified the is_our_turn function to make use of the new
num_available_members function to determine if it is our turn to try calling a member. There
is one small behavior change. Before writing this patch, if you had autofill disabled, then
if you were the head caller in a queue, you would automatically be told that it was your
turn to try calling a member. This did not take into account whether there were actually any
queue members available to take the call. Now we actually make sure there is at least one
member available to take the call if autofill is disabled.
(closes issue #13220)
Reported by: garychen
Review: http://reviewboard.digium.com/r/202/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After several issues raised on the Asterisk bugtracker against
the 1.4 branch were determined to be fixable with the state interface
change available in the 1.6.X series, it finally came time to just
suck it up and backport the change.
For a detailed explanation of what this change entails, the original
trunk commit for this feature may be found here:
http://svn.digium.com/view/asterisk?view=revision&revision=97203
In addition, the details for the use of this change to fix the problems
stated in issue #12970 may be found in the review request I made for
this change. It is linked below.
(closes issue #12970)
Reported by: edugs15
Review: http://reviewboard.digium.com/r/116
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
app_test was failing when sending the last DTMF digit, 8, because of the 100ms pause issued after DTMF is sent. During this pause the other side would hang up causing the test to look like it failed. Now the other side waits a second before hanging up.
(closes issue #12442)
Reported by: tzafrir
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184388 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The 'digit' variable is guaranteed to be non-NULL, so the if
statement could never evaluate true. Changing to ast_strlen_zero
makes the logic correct.
This was found while reviewing ast_channel_ao2 code review.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
issue #11583
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
With these changes, for a module to export symbols into the global namespace, it must have *both* the AST_MODFLAG_GLOBAL_SYMBOLS flag and a linker script that allows the linker to leave the symbols exposed in the module's .so file (see res_odbc.exports for an example).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@182802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue. Because using the ast prefix calls are
a better choice, ast_free_ptr is the new wrapper for free to pass to functions.
Also, a little bit of clean up was done to avoid the debug macros intentionally
being redefined.
(closes issue #13593)
Reported by: pj
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@181133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
