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2010-09-09Merged revisions 285710 via svnmerge from bbryant1-16/+53
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Formats need to load before apps, because some apps call ↵tilghman18-18/+90
ast_format_str_reduce() at load time. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279472 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Add load priority order, such that preload becomes unnecessary in most casestilghman18-107/+35
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16addition of G.719 pass-through supportdvossel1-0/+143
(closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270940 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-20Update supported file extensions in doxygen.lmadsen1-1/+1
Updated the doxygen \arg line after looking at the file for some other Asterisk documentation and noticing they weren't up to date. Thanks to seanbright for looking at the code for me :) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@257988 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-08Set a module load priority for format modules.russell17-18/+85
A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This fix applies to trunk, 1.6.1, and 1.6.2. The fix for 1.4 and 1.6.0 will require a different approach since the module load priority functionality is not present in the module API. (issue #16412) Reported by: jiddings git-svn-id: http://svn.digium.com/svn/asterisk/trunk@233692 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman17-57/+57
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Remove useless debugging message.kpfleming1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224562 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-15More 'static' qualifiers on module global variables.kpfleming1-1/+1
The 'pglobal' tool is quite handy indeed :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200620 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Const-ify the world (or at least a good part of it)kpfleming1-1/+1
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 186841 via svnmerge from mmichelson2-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines Fix a few typos of the word "frequency." (closes issue #14842) Reported by: jvandal Patches: frequency-typo.diff uploaded by jvandal (license 413) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186842 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Merged revisions 175825 via svnmerge from oej1-4/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175825 | oej | 2009-02-15 21:33:17 +0100 (Sön, 15 Feb 2009) | 2 lines format_ilbc does not depend on codec libraries and can therefore always be made. My mistake. Ursäkta! ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175827 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Merged revisions 175792 via svnmerge from oej1-1/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175792 | oej | 2009-02-15 21:20:21 +0100 (Sön, 15 Feb 2009) | 2 lines Disable format_ilbc.so by default, like codec_ilbc.so ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175801 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-13Add basic (passthrough, playback, record) support for ITU G.722.1 and ↵kpfleming2-0/+277
G.722.1C (also known as Siren7 and Siren14) This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported. Along the way, some related work was done: 1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way. 2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec. 3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result). Review: http://reviewboard.digium.com/r/158/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@175508 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-08Add alw as a valid file extension for alaw and ulw as a valid file extension ↵file1-2/+2
for ulaw. (closes issue #14001) Reported by: henrikw Patches: alw.diff uploaded by henrikw (license 627) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161869 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 ↵kpfleming4-8/+23
branch, and add the ones needed for all the new code here too git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153616 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Reverting format addition for nowtilghman1-71/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148071 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Add native 16kHz format for wav file format.tilghman1-12/+71
(Closes issue #13657) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148069 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-22Merged revisions 143903 via svnmerge from seanbright1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r143903 | seanbright | 2008-09-22 18:49:00 -0400 (Mon, 22 Sep 2008) | 8 lines Use the advertised header size in .au files instead of just assuming they are 24 bytes (the minimum). (closes issue #13450) Reported by: jamessan Patches: pcm-header.diff uploaded by jamessan (license 246) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143904 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07More merges from resolve-shadow warnings:seanbright1-34/+34
utils/ codecs/ and a change I missed from formats/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136408 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Start moving in changes from my resolve-shadow-warnings branch. Going to doseanbright2-105/+105
this in pieces so the diffs are a little bit smaller and more reviewable. pbx/ and formats/ first. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136298 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-08Janitor project to convert sizeof to ARRAY_LEN macro.bbryant1-2/+2
(closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129045 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22Fix a few places where frame data was used directly.qwell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117828 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causemvanbaak14-34/+34
- 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
2008-04-03Use a 32k file buffer on recordings, which increases the efficiency of file ↵tilghman1-1/+5
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
2008-03-28Merged revisions 111658 via svnmerge from qwell1-12/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) | 8 lines The file size of WAV49 does not need to be an even number. (closes issue #12128) Reported by: mdu113 Patches: 12128-noevenlength.diff uploaded by qwell (license 4) Tested by: qwell, mdu113 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@111659 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Merge changes from team/russell/g722-sillyness ...russell1-8/+19
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
2008-01-10minor formatting changesrussell1-8/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97804 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-07add a file-format driver for 16KHz signed linear... which may or may not workkpfleming1-0/+134
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96862 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-17Merged revisions 93180 via svnmerge from kpfleming1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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
2007-12-10Put into Makefile.moddir_rules the common instructions used torizzo1-9/+1
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
2007-12-09normalize subdirs' Makefile by using ASTTOPDIR and not .. to referencerizzo1-1/+1
the top level directory. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92022 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-29Merged revisions 90155 via svnmerge from tilghman7-20/+20
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90158 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-23formatting cleanup on the header,rizzo1-7/+7
normalization of the assignment of descriptor fields. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89530 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22remove a number of #include <fcntl.h> which are eitherrizzo1-1/+0
useless or done elsewhere git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89516 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22implement the split of file.h and mod_format.hrizzo14-14/+14
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89515 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove a bunch of useless #include "options.h"rizzo2-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89511 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19format handlers don't need network, lock, channel and scheduler headersrizzo14-97/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89427 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo14-28/+0
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
2007-11-16Start untangling header inclusion in a way that does not affectrizzo14-59/+1
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
2007-10-31More changes to change return values from load_module functions.qwell14-16/+42
(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
2007-06-29Merged revisions 72705 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r72705 | russell | 2007-06-29 15:56:18 -0500 (Fri, 29 Jun 2007) | 1 line give format_pcm a more concise destription ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72706 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-14oops... somebody patched this module without compile-testing it... bad :-)kpfleming1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69394 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.russell3-7/+5
(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
2007-04-08Merged revisions 60712 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60712 | tilghman | 2007-04-08 09:12:00 -0500 (Sun, 08 Apr 2007) | 2 lines Fix --enable-dev-mode ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60714 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06removed GAIN preprocessor definition, removed needsgain from struct ↵dhubbard1-41/+13
wav_desc, removed unnecessary gain code from wav_read() and wav_write() git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60641 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-05Merged revisions 60325 via svnmerge from dhubbard1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60325 | dhubbard | 2007-04-05 17:58:01 -0500 (Thu, 05 Apr 2007) | 1 line modified default GAIN for issue 5823, thanks jrwalliker ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60326 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-30Issue 8246 Doxygen updates (kshumard) oej1-4/+5
THANK YOU! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46434 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-25Merged revisions 46154 via svnmerge from kpfleming1-5/+21
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) | 2 lines add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46155 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-25This tiny fix prevents asterisk from crashing if trying to play an OGG moh file.murf1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43604 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-01 14:10:59 +0000