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features that will always be present in DAHDI
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) | 7 lines
Detect when sox fails to raise the volume, because sox can't read the file.
(closes issue #12939)
Reported by: rickbradley
Patches:
20080728__bug12939.diff.txt uploaded by Corydon76 (license 14)
Tested by: rickbradley
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(closes issue #13134)
Reported by: eliel
Patches:
app_image.c.patch uploaded by eliel (license 64)
UPGRADE.patch uploaded by eliel (license 64)
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(closes issue #13155)
Reported by: greenfieldtech
Patches:
app_voicemail.c.patch uploaded by greenfieldtech (license 369)
hebrew.ods uploaded by greenfieldtech (license 369)
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Pointed out by Atis Lezdins in #asterisk-dev
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r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines
As suggested by seanbright, the PSEUDO_CHAN_LEN in
app_chanspy should be set at load time, not at compile
time, since dahdi_chan_name is determined at load time.
Also changed the next_unique_id_to_use to have the
static qualifier.
Also added the dahdi_chan_name_len variable so that
strlen(dahdi_chan_name) isn't necessary. Thanks to
seanbright for the suggestion.
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r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines
Zap/pseudo is ten characters, but DAHDI/pseudo is
twelve. The strncmp call in next_channel should
account for this.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines
Update the "last" channel in next_channel in app_chanspy so
that the same pseudo channel isn't constantly returned.
related to issue #13124
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supported on a channel (yet _another_ useful patch by eliel).
(closes issue #13081)
Reported by: eliel
Patches:
app_sendtext.c.patch uploaded by eliel (license 64)
Tested by: eliel
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own repository, and must be installed like any other library for Asterisk to
use. The two modules that require it are codec_resample and app_jack.
To install libresample:
$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install
This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.
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impact on my machine ..
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the SendDTMF application. Also correct the default
pause between digits.
(closes issue #13102)
Reported by: eliel
Patches:
app_senddtmf.c.patch uploaded by eliel (license 64)
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(closes issue #13082)
Reported by: eliel
Patches:
app_rpt.c.patch uploaded by eliel (license 64)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines
Move the init_queue call back to where it used to be (changed
Sept 12 last year). It was moved then to prevent a memory leak.
Since then, the same memory leak recurred and was fixed in a
better way.
Now it has been found that the placement of this init_queue
call can cause problems if a realtime queue has values changed
to an empty string. The problem is that the default value
for that queue parameter would not be set.
(closes issue #13084)
Reported by: elbriga
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r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul 2008) | 6 lines
Apparently, "thread safety" is important, whatever
that means. :P
(Thanks Russell!)
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r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines
Make absolutely certain that the transfer datastore
is removed from the calling channel once the caller
is finished in the queue. This could have weird con-
sequences when dialing local queue members when multiple
transfers occur on a single call.
Also fixed a memory leak that would occur when an
attended transfer occurred from a queue member.
(closes issue #13047)
Reported by: festr
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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a member of any queue.
(closes issue #13073)
Reported by: eliel
Patches:
app_queue.c.patch uploaded by eliel (license 64)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines
Add a check to the CAN_EARLY_BRIDGE macro in app_dial to
be sure there are no audiohooks present on the channels
involved. This fixed a one-way audio situation I had in
my test setup. I couldn't find any open issues that suggested
one-way audio with regards to mixmonitor (or other audiohook)
usage, though.
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(closes issue #13054)
Reported by: pabelanger
Patches:
ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright
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fn2 was used in three functions. In every case, it was initialized
in the function it was used in. This meant there was no need
to have it in a malloc'd structure just taking up space. Furthermore
two of the functions it was used in were completely unnecessary since
fn2 was set to exactly the same value as the vm_state's fn string.
fn2 was a char array sized at PATH_MAX. On my system, PATH_MAX is
4096. This equates to a 4K memory savings per vm_state allocated.
Since there is a vm_state malloc'd for every voicemail user on
the system, this could potentially add up nicely if there are lots
of users. In addition, a vm_state is allocated on the stack each
time a caller calls the VoiceMailMain application, meaning that
there is a significant stack savings with this patch too.
Of course, a single vm_state struct still takes up approximately
20K on my system (when using IMAP storage. Without IMAP storage,
there would be about another 300 bytes fewer usage), even with
this removal. Further optimizations are probably possible,
but most likely not as easy as this one.
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state_interface, if a state_interface was set, on reload because the
state_interface isn't stored in the ast_db.
(closes issue #13043)
Reported by: jvandal
Patches:
app_queue.patch uploaded by jvandal (license 413)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines
Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not
registered.
(closes issue #12885)
Reported by: ibc
Patches:
20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: ibc
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(closes issue #13002)
Reported by: caio1982
Patches:
janitor_arraylen5.diff uploaded by caio1982 (license 22)
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t38_terminal_release, and make sure that the phase E handler gets called
with proper status.
(closes issue #13020)
Reported by: dimas
Patches:
v1-appfax.patch uploaded by dimas (license 88)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) | 7 lines
Check for non-NULL before stripping characters.
(closes issue #12954)
Reported by: bfsworks
Patches:
20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
Tested by: deti
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r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) | 2 lines
Stop using deprecated method, as requested by Kevin.
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app_voicemail would attempt to play a file called vm-foo instead of playing
vm-INBOX to play the "new" sound file. This commit fixes that issue.
This may fix one of the problems reported in issue #12987
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Brought up by reporter on issue #12987
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r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul 2008) | 6 lines
a couple of small Solaris-related fixes
(closes issue #11885)
Reported by: snuffy, asgaroth
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r127895 | kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 lines
remove this, it has been moved to the main Makefile
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of the whisper or barge audiohooks fails.
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(closes issue #12986)
Reported by: andrew53
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explanation of the change may be found in configs/queues.conf.sample
(closes issue #12690)
Reported by: atis
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(Closes issue #12892)
Reported by: jaroth
Patch originally by jaroth, fixed by me.
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r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul 2008) | 5 lines
Add error message to failed open(2) calls inside the copy() function of
app_voicemail. This idea came as part of my work in helping to resolve
issue #12764.
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'dialcontext' is only going to be set once during the channel's lifetime, so make it a string field instead of a char array
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Thanks to Russell for pointing out the problem
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MixMonitor to mix audio. However, it was pointed out to me that because
of this, the command set for the MONITOR_EXEC variable is ignored as well.
This means that people can't do their own custom mixing commands at the end
of recordings in order to make, for instance, stereo recordings of calls.
With this patch, app_queue will set the "joinfiles" variable for the channel's
monitor if MONITOR_EXEC is not zero-length. This means that for normal audio
mixing, MixMonitor is still the preferred choice, but we allow custom
mixing to be done with the two Monitor streams if desired.
(closes issue #12923)
Reported by: snyfer
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to how language is handled between two channels whose native
language is different. Prior to this patch, app_dial would have
the callee inherit the caller's language, and app_queue would not.
After this patch, app_dial no longer has the language inheritance
capability. This seems to make the most sense since it seems more
natural for a person to hear files played back in his/her native
language instead of the language of the person on the far end of
the call. See the CHANGES file for hints on how to keep the
previous behavior of app_dial if desired.
(closes issue #12489)
Reported by: bcnit
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