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anything with it.
(closes issue #13955)
Reported by: chris-mac
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@158851 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/trunk
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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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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.
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branch, and add the ones needed for all the new code here too
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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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audio files.
(closes issue #13372)
Reported by: epicac
Patches:
13372.patch uploaded by putnopvut (license 60)
Tested by: epicac
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140433 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137082 f38db490-d61c-443f-a65b-d21fe96a405b
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- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122523 f38db490-d61c-443f-a65b-d21fe96a405b
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117802 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines
Only try to prefix language if we are not using an absolute path (suffix it otherwise).
en/var/lib/asterisk/sounds/blah.gsm is a very silly path.
(closes issue #12379)
Reported by: kuj
Patches:
12379-absolutepath.diff uploaded by qwell (license 4)
Tested by: kuj, qwell
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recording.
(closes issue #11962)
Reported by: garlew
Patches:
recording.patch uploaded by garlew (license 376)
bug-11962.diff uploaded by snuffy (license 35)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@112564 f38db490-d61c-443f-a65b-d21fe96a405b
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rate
other than 8 kHz. The issue here is that format modules give a "whennext" sample
value, which is used to calculate when to set a timer for to retrieve the next
frame. However, the zaptel timer operates on 8 kHz samples, so this must be taken
into account.
(another part of issue #12164, reported by milazzo and jsmith, patch by me)
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Fix a number of other places where the number of samples in a G722 frame was
not properly handled because of various reasons.
main/rtp.c:
- When a G722 frame is read from the smoother, the number of samples in the
frame must be divided by 2 before being sent out over the network. Even
though G722 is 16 kHz, an error in some previous spec has made it so that
we have to list the number of samples such as if it was 8 kHz.
main/file.c:
- When scheduling the next time to expect a frame, take into account that the
format of the file we're reading from may not be 8 kHz.
codecs/codec_g722.c:
- When converting from G722 to slinear, g722_decode() expects its samples
parameter to be in the silly (real samples / 2) format. Make it so.
- When converting from slinear to G722, properly set the number of samples in
the frame to be the number of bytes of output * 2.
formats/format_pcm.c:
- This format module handles G722, among a number of other formats. However,
the read() and seek() functions did not account for the fact that G722 has
2 samples per byte.
(closes issue #12130, reported by rickross, patched by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106501 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12115)
Reported by: pj
Patches:
v2-fileexists.patch uploaded by dimas (license 88) (with modifications by me)
Tested by: dimas, qwell, russell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106439 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines
Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb 2008) | 4 lines
Bump a couple of more buffers up by 2 so that annoying warnings aren't generated
like crazy on every fileexists_core call.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2 lines
Bump up the buffer by 2.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb 2008) | 8 lines
fallback to standard English prompts properly when using new prompt directory layout
(closes issue #11831)
Reported by: IgorG
Patches:
fallbacken.v1.diff uploaded by IgorG (license 20) (modified by me to improve code and conform rest of function to coding guidelines)
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does not exist, not if the file itself could not be opened.
(closes issue #11828)
Reported by: IgorG
Patches:
readfile.v1.diff uploaded by IgorG (license 20)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103827 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008) | 12 lines
Fix a couple of places where ast_frfree() was not called on a frame that came
from a translator. This showed itself by g729 decoders not getting released.
Since the flag inside the translator frame never got unset by freeing the frame
to indicate it was no longer in use, the translators never got destroyed, and
thus the g729 licenses were not released.
(closes issue #11892)
Reported by: xrg
Patches:
11892.diff uploaded by russell (license 2)
Tested by: xrg, russell
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r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) | 11 lines
When deleting a task from the scheduler, ignoring the return value could
possibly cause memory to be accessed after it is freed, which causes all
sorts of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our taskid
value).
(closes issue #11386)
Reported by: flujan
Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007) | 5 lines
Use of "private" as a field name in a header file messes with C++ projects
Reported by: chewbacca
Patch by: casper
(Closes issue #11401)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | 4 lines
Merge a change from team/russell/chan_refcount ...
This makes ast_stopstream() thread-safe.
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who really need it.
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build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
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- the *_CURRENT macros no longer need the list head pointer argument
- add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists
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- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #10724)
Reported by: eliel
Patches:
chan_skinny.c.patch uploaded by eliel (license 64)
chan_oss.c.patch uploaded by eliel (license 64)
chan_mgcp.c.patch2 uploaded by eliel (license 64)
pbx_config.c.patch uploaded by seanbright (license 71)
iax2-provision.c.patch uploaded by eliel (license 64)
chan_gtalk.c.patch uploaded by eliel (license 64)
pbx_ael.c.patch uploaded by seanbright (license 71)
file.c.patch uploaded by seanbright (license 71)
image.c.patch uploaded by seanbright (license 71)
cli.c.patch uploaded by moy (license 222)
astobj2.c.patch uploaded by moy (license 222)
asterisk.c.patch uploaded by moy (license 222)
res_limit.c.patch uploaded by seanbright (license 71)
res_convert.c.patch uploaded by seanbright (license 71)
res_crypto.c.patch uploaded by seanbright (license 71)
app_osplookup.c.patch uploaded by seanbright (license 71)
app_rpt.c.patch uploaded by seanbright (license 71)
app_mixmonitor.c.patch uploaded by seanbright (license 71)
channel.c.patch uploaded by seanbright (license 71)
translate.c.patch uploaded by seanbright (license 71)
udptl.c.patch uploaded by seanbright (license 71)
threadstorage.c.patch uploaded by seanbright (license 71)
db.c.patch uploaded by seanbright (license 71)
cdr.c.patch uploaded by moy (license 222)
pbd_dundi.c.patch uploaded by moy (license 222)
app_osplookup-rev83558.patch uploaded by moy (license 222)
res_clioriginate.c.patch uploaded by moy (license 222)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
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r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines
gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set of
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)
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r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | 11 lines
Fix an issue that can occur when you do an attended transfer to parking. If
you complete the transfer before the announcement of the parking spot finishes,
then the channel being parked will hear the remainder of the announcement.
These changes make it so that will not happen anymore.
Basically, res_features sets a flag on the channel is playing the announcement
to so that the file streaming core knows that it needs to watch out for a
channel masquerade, and if it occurs, to abort the announcement.
(closes BE-182)
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r78955 | file | 2007-08-10 11:15:53 -0300 (Fri, 10 Aug 2007) | 2 lines
Don't bother having the core pass through or emulate begin DTMF frames when in an ast_waitstream. It only cares about the end of DTMF.
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(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r71068 | qwell | 2007-06-22 10:00:30 -0500 (Fri, 22 Jun 2007) | 12 lines
Merged revisions 71065 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 lines
Fix a few silly usages of ast_playstream() - it only ever returns 0...
Issue 10035
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(issue #9983, eliel)
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(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
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r67924 | file | 2007-06-06 19:38:15 -0400 (Wed, 06 Jun 2007) | 2 lines
Properly handle cases where a stream can't be written to. (issue #9757 reported by junky)
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guidelines changes
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r63152 | oej | 2007-05-06 14:28:38 +0200 (Sun, 06 May 2007) | 2 lines
Stop the video stream when you stop playback of all streams for a call
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r60661 | tilghman | 2007-04-07 20:40:47 -0500 (Sat, 07 Apr 2007) | 10 lines
Merged revisions 60660 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60660 | tilghman | 2007-04-07 20:39:25 -0500 (Sat, 07 Apr 2007) | 2 lines
Bug 9486 - memory leak when opening a filestream
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