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r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others
mixed in here, as well.
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
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r304097 | seanbright | 2011-01-25 20:26:26 -0500 (Tue, 25 Jan 2011) | 19 lines
Merged revisions 304096 via svnmerge from
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r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan 2011) | 12 lines
Per the man page, setvbuf() must be called before any other operation on an open file.
We use setvbuf() to associate a buffer with a stream, but we have already written
to the open file. This works (by chance) on Linux, but fails on other platforms,
such as OpenSolaris.
(closes issue #16610)
Reported by: bklang
Patches:
setvbuf.patch uploaded by crjw (license 963)
Tested by: bklang, asgaroth, efutch
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r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) | 2 lines
Removal of unused variables so Asterisk will compile.
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r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) | 7 lines
Call execl() directly for a better solution for paths with spaces.
(closes issue #18600)
Reported by: ebroad
Patches:
20110111__issue18600__2.diff.txt uploaded by tilghman (license 14)
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r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) | 4 lines
Quote arguments, just in case there's a space in a pathname.
(Diagnosed by pabelanger on #asterisk-dev, fixed by me.)
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r290576 | tilghman | 2010-10-06 08:49:19 -0500 (Wed, 06 Oct 2010) | 15 lines
Merged revisions 290575 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010) | 8 lines
Allow streaming audio from a pipe.
(closes issue #18001)
Reported by: jamicque
Patches:
20100926__issue18001.diff.txt uploaded by tilghman (license 14)
Tested by: jamicque
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r286270 | oej | 2010-09-11 19:09:22 +0200 (Lör, 11 Sep 2010) | 18 lines
Merged revisions 286268 via svnmerge from
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r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör, 11 Sep 2010) | 11 lines
Merged revisions 286267 via svnmerge from
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r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 lines
Handle error response when we can't make file compatible
Review: https://reviewboard.asterisk.org/r/911/
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The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
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Suppress the warning about unexpected control subclass frames for
AST_CONTROL_CONNECTED_LINE, AST_CONTROL_REDIRECTING, and AST_CONTROL_AOC
in file.c:waitstream_core().
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r254451 | twilson | 2010-03-25 10:57:29 -0500 (Thu, 25 Mar 2010) | 2 lines
Handle new SRCCHANGE control message here too
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r238629 | mnicholson | 2010-01-08 13:20:44 -0600 (Fri, 08 Jan 2010) | 5 lines
Properly calculate the remaining space in the output string when reducing format strings.
(closes issue #16560)
Reported by: goldwein
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r232007 | russell | 2009-12-01 17:25:36 -0600 (Tue, 01 Dec 2009) | 2 lines
Fix a warning pointed out by buildbot.
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r231740 | mnicholson | 2009-12-01 09:34:57 -0600 (Tue, 01 Dec 2009) | 2 lines
Ignore unknown formats in ast_format_str_reduce() and return an error if no know formats are found.
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r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines
Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list.
(closes issue #15625)
Reported by: Shagg63
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines
Make filestream frame handling safer by isolating frames before returning them.
This patch is related to a number of issues on the bug tracker that show
crashes related to freeing frames that came from a filestream. A number of
fixes have been made over time while trying to figure out these problems, but
there re still people seeing the crash. (Note that some of these bug reports
include information about other problems. I am specifically addressing
the filestream frame crash here.)
I'm still not clear on what the exact problem is. However, what is _very_
clear is that we have seen quite a few problems over time related to unexpected
behavior when we try to use embedded frames as an optimization. In some cases,
this optimization doesn't really provide much due to improvements made in other
areas.
In this case, the patch modifies filestream handling such that the embedded frame
will not be returned. ast_frisolate() is used to ensure that we end up with a
completely mallocd frame. In reality, though, we will not actually have to malloc
every time. For filestreams, the frame will almost always be allocated and freed
in the same thread. That means that the thread local frame cache will be used.
So, going this route doesn't hurt.
With this patch in place, some people have reported success in not seeing the
crash anymore.
(SWP-150)
(AST-208)
(ABE-1834)
(issue #15609)
Reported by: aragon
Patches:
filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2)
Tested by: aragon, russell
(closes issue #15817)
Reported by: zerohalo
Tested by: zerohalo
(closes issue #15845)
Reported by: marhbere
Review: https://reviewboard.asterisk.org/r/386/
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r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines
Really stop the stream, when ast_closestream() is called.
(closes issue #15129)
Reported by: bmh
Patches:
20090918__issue15129.diff.txt uploaded by tilghman (license 14)
Review:
https://reviewboard.asterisk.org/r/372/
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This patch adds a document describing the language prompt submission process,
licensing terms and other issues related to that process. In addition, it
modifies the sound file searching process to support language codes with
any number of suffices (not limited to just "xx" or "xx_YY"), so that prompts
can be named with gender, customer/company, etc. suffices as well.
(closes issue #15771)
Reported by: jtodd
Patches:
language-criteria.txt uploaded by jtodd
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Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
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r203785 | russell | 2009-06-26 16:16:39 -0500 (Fri, 26 Jun 2009) | 15 lines
Don't fast forward past the end of a message.
This is nice change for users of the voicemail application. If someone gets a
little carried away with fast forwarding through a message, they can easily
get to the end and accidentally exit the voicemail application by hitting the
fast forward key during the following prompt.
This adds some safety by not allowing a fast forward past the end of a message.
(closes issue #14554)
Reported by: lacoursj
Patches:
21761.patch uploaded by lacoursj (license 707)
Tested by: lacoursj
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
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/
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The 'pglobal' tool is quite handy indeed :-)
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r188582 | mmichelson | 2009-04-15 15:04:20 -0500 (Wed, 15 Apr 2009) | 7 lines
Update ast_readvideo_callback to match ast_readaudio_callback.
This fixes potential refcount errors that may occur on ast_filestreams.
AST-208
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r187300 | tilghman | 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) | 3 lines
Add debugging mode for diagnosing file descriptor leaks.
(Related to issue #14625)
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r187301 | tilghman | 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) | 2 lines
Oops, missed this file in the last commit.
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was not installed.
(closes issue #14541)
Reported by: grant
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The problem came from the fact that a frame read from a format interpreter
was not freed. Adding a call to ast_frfree fixed this. The explanation for
why this caused the problem is a bit complex, but here goes:
There was a problem in all versions of Asterisk where the embedded frame
of a filestream structure was referenced after the filestream was freed. This
was fixed by adding reference counting to the filestream structure. The refcount
would increase every time that a filestream's frame pointer was pointing to an
actual frame of data. When the frame was freed, the refcount would decrease. Once
the refcount reached 0, the filestream was freed, and as part of the operation,
the open files were closed as well.
Thus it becomes more clear why a missing ast_frfree would cause a reference leak
and cause the files to not be closed. You may ask then if there was a frame leak
before this patch. The answer to that is actually no! The filestream code was
"smart" enough to know that since the frame we received came from a format interpreter,
the frame had no malloced data and thus didn't need to be freed. Now, however, there
is cleanup that needs to be done when we finish with the frame, so we do need to
call ast_frfree on the frame to be sure that the refcount for the filestream is
decremented appropriately.
(closes issue #14384)
Reported by: fiddur
Patches:
14384.patch uploaded by putnopvut (license 60)
Tested by: fiddur, putnopvut
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r167566 | russell | 2009-01-07 16:35:36 -0600 (Wed, 07 Jan 2009) | 2 lines
Fix the last couple of places where free() was improperly used directly.
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r167554 | russell | 2009-01-07 16:26:42 -0600 (Wed, 07 Jan 2009) | 2 lines
Don't fclose() the file early, the filestream destructor will handle it.
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r167545 | russell | 2009-01-07 16:19:47 -0600 (Wed, 07 Jan 2009) | 2 lines
Only try to close the file if one was actually opened
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r167541 | russell | 2009-01-07 16:03:59 -0600 (Wed, 07 Jan 2009) | 4 lines
Don't use free() directly. This caused a crash since ast_filestream is now an ao2 object.
Reported by JunK-Y on IRC, #asterisk-dev
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A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.
A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.
(closes issue #14118)
Reported by: blitzrage
Patches:
14118v2.patch uploaded by putnopvut (license 60)
Tested by: blitzrage
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of directly seeking to zero. This is because some audio formats have headers at the front that need to be skipped, which will be done by the format module.
(closes issue #14079)
Reported by: elguero
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(closes issue #13990)
Reported by: eliel
Patches:
array_len.diff uploaded by eliel (license 64)
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anything with it.
(closes issue #13955)
Reported by: chris-mac
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r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines
Begin on a crusade to end trailing whitespace!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158133 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reported by: snyfer
This handles the case for a zero length file to attempt to be streamed. Instead of failing from not playing any data, go ahead and return success as ast_streamfile should consider playing nothing a success when there is nothing to play.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158062 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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branch, and add the ones needed for all the new code here too
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@148070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r142740 | tilghman | 2008-09-12 11:27:32 -0500 (Fri, 12 Sep 2008) | 4 lines
Don't return a free'd pointer, when a file cannot be opened.
(closes issue #13462)
Reported by: wackysalut
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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audio files.
(closes issue #13372)
Reported by: epicac
Patches:
13372.patch uploaded by putnopvut (license 60)
Tested by: epicac
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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