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Author SHA1 Message Date
file
35d5a377ed Merged revisions 98951 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan 2008) | 4 lines

Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex.
(closes issue #11693)
Reported by: yzg

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98952 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16 01:17:25 +00:00
russell
b61a98675c Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines

Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.

The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed.  Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code.  The reason this
happens is that the channel might get masqueraded during this time.  During a
masquerade, existing translation paths get destroyed.

So, this patch fixes the issue in an API and ABI compatible way.  (This one is
 for you, paravoid!)

It changes an int in ast_frame to be used as flag bits.  The 1 bit is still used
to indicate that the frame contains timing information.  Also, a second flag has
been added to indicate that the frame came from a translator.  When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed.  At this point, the flag gets
cleared.  Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.

Admittedly, this feels like a hack.  But, it does fix the issue, and I was not able 
to think of a better solution ...

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98944 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-15 23:31:53 +00:00
russell
9c1c46c009 Kevin noted that the thing that I _actually_ changed here was that I converted
a value from a double, to a float, back to a double.  Sure enough, when I changed
my interim variable back to a double, it still blows up.  Switching all of these
to a float fixes the problem.  This seems like a compiler bug where a double passed
as an argument isn't getting properly aligned, so I'll have to see if I can replicate
it with a small test program.

(related to issue #11725)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98308 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 19:05:24 +00:00
russell
ccce263b5c Fix a bus error that happened when asterisk was built with optimizations on
with platforms that explode on unaligned access.  I'm not exactly sure why
this fixes it, but it fixed it on the machine I was testing on.  If it makes
sense to you, feel free to enlighten me.  :)

(closes issue #11725, patched by me)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98270 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 18:48:07 +00:00
russell
2f83bcc869 At one point during working on this module, I had the lin/lin16 versions of the
framein callbacks different.  However, they are now the same again, so remove
the duplicate code and use the same functions for the lin/lin16 versions.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 17:17:54 +00:00
russell
560327b0ec - Fix the last set of places where incorrect assumptions were made about the
sample length with g722.  It is _2_ samples per byte, not 1.  This was all
   over the place, and I believed it, and it is what caused me to take so long
   to figure out what was broken.
 - Update copyright information on codec_g722.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98081 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 03:37:19 +00:00
russell
300fa53d79 Fix various issues in codec_g722.
- The most common fix being made here is to fix all of the places where the
   number of output samples and output bytes gets updated in the translator
   state structure.
 - Fix a number of other places where the number of samples provided as an
   initialization value to a struct was incorrect.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97975 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10 23:16:09 +00:00
russell
57ccc02998 Fix the buffer_samples value. For signed linear, the number of samples needed
to fill the buffer is half the buffer size.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97974 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10 23:10:00 +00:00
russell
a2a1eb045d Fix this so it doesn't force codec_g722 to get relinked every time
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97652 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10 00:17:02 +00:00
russell
d4f2402f2e Ensure that libg722.a gets rebuilt if one of the files changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97650 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10 00:11:02 +00:00
kpfleming
bf8028a2c8 Merged revisions 97491 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008) | 2 lines

report the same message whether Zaptel does not have transcoder support loaded or no transcoders were found

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97495 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09 17:30:13 +00:00
kpfleming
3f0b6d8d86 and now just to keep the libresample party going... if the functions from libresample are going to be in the main Asterisk binary, it makes sense for the header that defines them to be available without any special CFLAGS and to out-of-tree modules building against /usr/include/asterisk
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95894 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02 18:21:04 +00:00
kpfleming
933ddf410a go back to including libresample in the main Asterisk binary, but this time including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_<foo> symbols alone)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95816 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02 14:05:30 +00:00
russell
21969815a0 Instead of linking libresample into the main Asterisk binary, build it as
res_resample, and mark codec_resample as dependent upon res_resample.  This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places.  (I have another module
in a branch that needs it, too.)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95697 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02 01:00:44 +00:00
file
59bc75d1f8 Fix building of codec_resample on platforms other then Cygwin. On everything else it actually gets built after codec_resample, so you can't exactly link it in since it doesn't exist.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95648 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01 23:09:32 +00:00
rizzo
5503f69d4c make codec_resample build on __CYGWIN__, and make it load on FreeBSD
(and probably other systems as well).
Both need libresample.a to be specified in the linking phase,
and cygwin needs <float.h> as other BSD.

The checks for OS-specific headers should really be moved to some
common header though.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95625 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-01 22:21:39 +00:00
russell
250f46db38 Use float.h to fix the build on FreeBSD. Also, add some other platforms as
they are likely the same.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95550 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31 22:41:39 +00:00
russell
04838b9d59 Merge changes from team/russell/codec_resample
This commit imports libresample for use in Asterisk.  It also adds a new codec
module, codec_resample.  This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.

It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz 
signed linear.  But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95501 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31 21:22:31 +00:00
russell
806b3cfdd3 I went looking for where we downloaded the g722 implementation and came across
these two links.  So, I'm adding them so they are available for reference later.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94877 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27 16:11:41 +00:00
qwell
7c0eb401e6 codecs.conf really shouldn't be mandatory.. it never had been before, so let's go back to being optional.
A big "thank you" to pnlarsson on IRC for allowing me access to his system to debug this.

Closes issue #11584.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94541 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21 20:12:50 +00:00
kpfleming
d4e966efcc Merged revisions 93180 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines

In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html,
rizzo brought up some issues related to the way that the metadata required
for menuselect and the rest of the build system is extracted from the source
files. Since I had a few hours to kill on an airplane today, I decided to
improve this situation... so now the system caches the extracted metadata
and uses it to build the menuselect 'tree' as much as it can. The result
of this is that when a single source file is changed, only the metadata for
that file needs to be extracted again, and the rest is used from the cache
files. I also reduced the number of forked processes required to do the
metadata extraction; it was actually possible to do most of what we needed
in the Makefiles themselves without using any shell scripts at all! On my
laptop, these changes resulted in an 80% decrease in the time required
for the 'menuselect.makeopts' automatic check to occur after editing a single
source file.

While doing this work I also cleaned up a few minor things in the Makefiles,
adding a check for 'awk' to the configure script and changed all remaining
places we use 'grep' or 'awk' to use the ones found by the configure script,
and changed the 'prep_tarball' script to build the menuselect metadata so
that tarballs of Asterisk will include it and won't require the user to
wait while it is extracted after unpacking.


........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17 07:25:35 +00:00
tilghman
a4425cc28d Solaris compat fixes
Reported by: snuffy
Patch by: snuffy,tilghman
(Closes issue #11315)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93090 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14 21:09:17 +00:00
rizzo
aa85540763 Put into Makefile.moddir_rules the common instructions used to
generate loadable and embedded module lists.

Individual Makefiles now are a lot simpler, possibly as simple as this:

    -include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
    MODULE_PREFIX=cdr_
    all: _all
    include $(ASTTOPDIR)/Makefile.moddir_rules

and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.

The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).

With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.




git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92082 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-10 03:50:38 +00:00
rizzo
b50ce18fe8 normalize subdirs' Makefile by using ASTTOPDIR and not .. to reference
the top level directory.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92022 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-09 21:29:37 +00:00
rizzo
8cd33321ef remove a number of #include <fcntl.h> which are either
useless or done elsewhere



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89516 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 01:03:02 +00:00
rizzo
e8c3c0d206 remove some useless includes from codecs
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89428 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 19:51:55 +00:00
rizzo
9cf442d7f7 include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere.
Remove some of the instances.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 18:52:04 +00:00
rizzo
883346d64a Start untangling header inclusion in a way that does not affect
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.).



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16 20:04:58 +00:00
kpfleming
a45a413db3 improve linked-list macros in two ways:
- 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


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89106 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-08 05:28:47 +00:00
tilghman
4b2fc9d3e7 Commit some cleanups to the format type code.
- 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.)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89071 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06 22:51:48 +00:00
qwell
6314703e32 Merged revisions 89046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4 lines

Correctly set the total number of channels from a zaptel transcoder board.

SPD-49, patch by Matthew Nicholson.

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89047 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06 19:10:18 +00:00
qwell
9e15a0e72b More changes to change return values from load_module functions.
(issue #11096)
Patches:
      codec_adpcm.c.patch uploaded by moy (license 222)
      codec_alaw.c.patch uploaded by moy (license 222)
      codec_a_mu.c.patch uploaded by moy (license 222)
      codec_g722.c.patch uploaded by moy (license 222)
      codec_g726.c.diff uploaded by moy (license 222)
      codec_gsm.c.patch uploaded by moy (license 222)
      codec_ilbc.c.patch uploaded by moy (license 222)
      codec_lpc10.c.patch uploaded by moy (license 222)
      codec_speex.c.patch uploaded by moy (license 222)
      codec_ulaw.c.patch uploaded by moy (license 222)
      codec_zap.c.patch uploaded by moy (license 222)
      format_g723.c.patch uploaded by moy (license 222)
      format_g726.c.patch uploaded by moy (license 222)
      format_g729.c.patch uploaded by moy (license 222)
      format_gsm.c.patch uploaded by moy (license 222)
      format_h263.c.patch uploaded by moy (license 222)
      format_h264.c.patch uploaded by moy (license 222)
      format_ilbc.c.patch uploaded by moy (license 222)
      format_jpeg.c.patch uploaded by moy (license 222)
      format_ogg_vorbis.c.patch uploaded by moy (license 222)
      format_pcm.c.patch uploaded by moy (license 222)
      format_sln.c.patch uploaded by moy (license 222)
      format_vox.c.patch uploaded by moy (license 222)
      format_wav.c.patch uploaded by moy (license 222)
      format_wav_gsm.c.patch uploaded by moy (license 222)
      res_adsi.c.patch uploaded by eliel (license 64)
      res_ael_share.c.patch uploaded by eliel (license 64)
      res_clioriginate.c.patch uploaded by eliel (license 64)
      res_convert.c.patch uploaded by eliel (license 64)
      res_indications.c.patch uploaded by eliel (license 64)
      res_musiconhold.c.patch uploaded by eliel (license 64)
      res_smdi.c.patch uploaded by eliel (license 64)
      res_speech.c.patch uploaded by eliel (license 64)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87889 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31 19:24:29 +00:00
kpfleming
5aafb5a5e4 clean up assembler and preprocessor files if they are here too
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87467 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-29 22:24:44 +00:00
qwell
7756b987a0 Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22 20:05:18 +00:00
qwell
d542122e6a Convert NEW_CLI to AST_CLI.
Closes issue #11039, as suggested by seanbright.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19 18:29:40 +00:00
qwell
4723d35127 More changes to NEW_CLI.
Also fixes a few cli messages and some minor formatting.

(closes issue #11001)
Reported by: seanbright
Patches:
      newcli.1.patch uploaded by seanbright (license 71)
      newcli.2.patch uploaded by seanbright (license 71)
      newcli.4.patch uploaded by seanbright (license 71)
      newcli.5.patch uploaded by seanbright (license 71)
      newcli.6.patch uploaded by seanbright (license 71)
      newcli.7.patch uploaded by seanbright (license 71)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86534 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19 18:01:00 +00:00
russell
a51f0482f6 Merged revisions 86296 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r86296 | russell | 2007-10-18 10:45:55 -0500 (Thu, 18 Oct 2007) | 3 lines

Execute the RELEASE operation on transcoder channels in the destroy callback.
(patch from jsloan)

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86297 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-18 15:57:30 +00:00
russell
08c7dd2cbd The trunk version of this patch also includes a couple more small clean fixes
from IgorG.

Merged revisions 84170 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r84170 | russell | 2007-10-01 10:00:56 -0500 (Mon, 01 Oct 2007) | 3 lines

Remove another file in "make clean".
(closes issue #10814, paravoid)

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84171 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-01 15:06:14 +00:00
file
e4953f8f10 Merged revisions 82265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r82265 | file | 2007-09-11 18:41:49 -0300 (Tue, 11 Sep 2007) | 4 lines

(closes issue #10679)
Reported by: andrew
Build under dev mode when K6OPTS is enabled.

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82266 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-11 21:43:47 +00:00
kpfleming
44f5ebb9b4 Merged revisions 81405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r81405 | kpfleming | 2007-08-31 10:51:45 -0500 (Fri, 31 Aug 2007) | 2 lines

add missing "transcoder show" (and deprecated "show transcoder") CLI commands that were in 1.2 but never added to 1.4

........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81408 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-31 15:58:31 +00:00
murf
8b5681e22a This change set fixes bug 8126 in trunk. It is implemented via compile time options, activated via the menuselect stuff, which defaults to the old way. non-zero sample data added. Translate tables expressed in microseconds instead of milliseconds, with 5-digit data now instead of 3, giving 2 more digits of precision.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@80113 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-20 22:53:48 +00:00
tilghman
dbec3d56c1 Don't reload a configuration file if nothing has changed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16 21:09:46 +00:00
russell
4f3c4dc7f2 Do a massive conversion for using the ast_verb() macro
(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");


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77299 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-26 15:49:18 +00:00
russell
f042431847 Add a massive set of changes for converting to use the ast_debug() macro.
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69327 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-14 19:39:12 +00:00
tilghman
eb5d461ed4 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06 21:20:11 +00:00
qwell
943e4bad3d Merged revisions 65877 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65877 | qwell | 2007-05-24 11:14:02 -0400 (Thu, 24 May 2007) | 4 lines

Fix handling of zero-length frames when a codec is capable of native PLC.

Issue 9183, patch by Mihai.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65903 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24 15:28:29 +00:00
file
d185cf4928 Merged revisions 64278 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64278 | file | 2007-05-14 14:48:33 -0400 (Mon, 14 May 2007) | 2 lines

Properly set datalen field when doing PLC in codec_speex. (issue #9722 reported by mihai)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64279 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-14 18:49:40 +00:00
qwell
c648b0d70f Merged revisions 62174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62174 | qwell | 2007-04-27 11:17:46 -0500 (Fri, 27 Apr 2007) | 11 lines

Merged revisions 62173 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62173 | qwell | 2007-04-27 11:16:16 -0500 (Fri, 27 Apr 2007) | 3 lines

This transcoder message needn't be a NOTICE.
I've seen it cause confusion more than a few times.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62175 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-27 16:18:51 +00:00
kpfleming
7337a7aeb1 Merged revisions 60399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r60399 | kpfleming | 2007-04-06 09:49:51 -0500 (Fri, 06 Apr 2007) | 10 lines

Merged revisions 60398 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r60398 | kpfleming | 2007-04-06 09:41:37 -0500 (Fri, 06 Apr 2007) | 2 lines

remove undocumented 'cardsmode' parameter and stop searching for transcoders during reload()

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60400 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06 14:53:13 +00:00
russell
14465f3016 Sync codec_zap with the one that is in the 1.4 branch so that it can actually
build here, too.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58101 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06 22:15:02 +00:00