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2011-07-14Merged revisions 328247 via svnmerge from lmadsen11-0/+36
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2011-07-05Merged revisions 326411 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) | 14 lines Add the attribute "type" to each "<use>" for menuselect. This matters only when autoconf fails to detect that weak linking is supported. External optional dependencies will become optional in both cases, as they are removed at compile time when not detected. However, runtime-optional modules are made mandatory when weak linking is not found. This change affects only the external optional dependencies; previously, they were incorrectly required when weak linking support was not detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003) Tested by: iasgoscouk ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@326412 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-03Merged revisions 316265 via svnmerge from russell1-2/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ 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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316293 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-20codec_dahdi: DAHDI still advertises formats using the old bitfields.sruffell1-6/+20
Previously, the DAHDI format bit fields matched up with the Asterisk bitfields. Since the Asterisk codec bit fields were replaced in r306010, codec_dahdi needs to contain the formats itself. In the future, the DAHDI formats should either change to something other than bitfields, or the bitfields need to move from include/dahdi/kernel.h to include/dahdi/user.h. Signed-off-by: Shaun Ruffell <sruffell@digium.com> git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314471 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-20Fixes error with frame datalen being calculated from samples when this is ↵dvossel1-1/+6
not allwaya accurate. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314415 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-18Remove libresample dependency from codec_resample.cdvossel1-4/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@311385 f38db490-d61c-443f-a65b-d21fe96a405b
2011-03-11Use "-march=native" when possible.kpfleming2-36/+46
Recent versions of GCC have a tuning option value of 'native', which causes the compiler to optimize the build for the CPU the compile is performed on. Since most people are building Asterisk on the machine they plan to run it on, the configure script and build system will now use this value unless a different value is specified by the user in CFLAGS when the configure script is executed. In addition, this value will be used for building the GSM and LPC10 codecs as well, in preference to the logic that has been in their Makefiles forever to optimize for certain types of CPUs. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@310332 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-22Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd ↵dvossel9-143/+2176
audio ConfBridge, and other stuff -Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308582 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Replace ast_log(LOG_DEBUG, ...) with ast_debug()pabelanger1-4/+4
(closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306258 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Fix compile error in codec ilbc translator.dvossel2-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306257 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Asterisk media architecture conversion - no more format bitfieldsdvossel20-93/+137
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/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-05Merged revisions 293970 via svnmerge from sruffell1-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293970 | sruffell | 2010-11-04 19:07:11 -0500 (Thu, 04 Nov 2010) | 32 lines Merged revisions 293969 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500 (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically commit 9034) added the capability for the wctc4xxp to return more than a single packet of data in response to a read. However, when decoding packets, codec_dahdi was still assuming that the default number of samples was in each read. In other words, each packet your provider sent you, regardless of size, would result in 20 ms of decoded data (30 ms if decoding G723). If your provider was sending 60 ms packets then codec_dahdi would end up stripping 40 ms of data from each transcoded frame resulting in "choppy" audio. This would only affect systems where G729 packets are arriving in sizes greater than 20ms or G723 packets arriving in sizes greater than 30ms. DAHDI-744. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@293971 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Merged revisions 285819 via svnmerge from pabelanger1-0/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285819 | pabelanger | 2010-09-09 18:52:31 -0400 (Thu, 09 Sep 2010) | 22 lines Merged revisions 285818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400 (Thu, 09 Sep 2010) | 15 lines Merged revisions 285817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines GCC 4.2.x optimizations result in improper behavior of GSM codec (closes issue #17688) Reported by: pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347) Tested by: mkeuter, pprindeville ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@285820 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-21add speex 16khz sample frame so codec cost can be calculateddvossel2-5/+36
(closes issue #17534) Reported by: fabled Patches: speex-wb-sample.diff uploaded by fabled (license 448) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271625 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17adds speex 16khz audio supportdvossel1-17/+65
(closes issue #17501) Reported by: fabled Patches: asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448) Tested by: malcolmd, fabled, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271231 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Make compile again.rmudgett1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267622 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Remove unnecessary code relating to PLC.mmichelson8-215/+0
The logic for handling generic PLC is now handled in ast_write in channel.c instead of in translation code. Review: https://reviewboard.asterisk.org/r/683/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267492 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Change per-file debug and verbose levels to be per-module, the waykpfleming1-0/+1
users expect them to work. 'core set debug' and 'core set verbose' can optionally change the level for a specific filename; however, this is actually for a specific source file name, not the module that source file is included in. With examples like chan_sip, chan_iax2, chan_misdn and others consisting of multiple source files, this will not lead to the behavior that users expect. If they want to set the debug level for chan_sip, they want it set for all of chan_sip, and not to have to also set it for reqresp_parser and other files that comprise the chan_sip module. This patch changes this functionality to be module-name based instead of file-name based. To make this work, some Makefile modifications were required to ensure that the AST_MODULE definition is present in each object file produced for each module as well. Review: https://reviewboard.asterisk.org/r/574/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253917 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-16OSARCH is not inherited to this directorytilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252760 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-15Make the Makefile logic more explicit and move the Snow Leopard logic down ↵tilghman1-10/+12
to where it's not executed on non-Darwin systems. (closes issue #17028) Reported by: pabelanger Patches: issue17028_20100315.patch uploaded by seanbright (license 71) 20100315__issue17028.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252488 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-09Build system modifications to ensure that Asterisk properly builds on Mac OS ↵tilghman1-0/+5
X 10.6. (closes issue #16997) Reported by: jquinn Patches: 20100309__issue16997__2.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251475 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-10Merged revisions 229281 via svnmerge from file1-42/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r229281 | file | 2009-11-10 16:03:14 -0400 (Tue, 10 Nov 2009) | 8 lines Remove broken support for direct transcoding between G.726 RFC3551 and G.726 AAL2. On some systems the translation core would actually consider g726aal2 -> g726 -> signed linear to be a quicker path then g726aal2 -> signed linear which exposed this problem. (closes issue #15504) Reported by: globalnetinc ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@229282 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Fixes merging issue from 1.4, frame data is held in data.ptr in trunkdvossel1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228441 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Merged revisions 228418 via svnmerge from dvossel1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009) | 13 lines fixes segfault in iLBC For reasons not yet known, it appears possible for an ast_frame to have a datalen greater than zero while the actual data is NULL during Packet Loss Concealment. Most codecs don't support PLC so this doesn't affect them. This patch catches the malformed frame and prevents the crash from occuring. Additional efforts to determine why it is possible for a frame to look like this are still being investigated. (issue #16979) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@228420 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman11-17/+42
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 224931 via svnmerge from russell1-8/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224932 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Fixes numerous spelling errors. Patch submitted by alecdavis.dbrooks1-1/+1
(closes issue #15595) Reported by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209554 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-21Merged revisions 207647 via svnmerge from kpfleming3-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are honored. This commit changes the build system so that user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided by the build system itself, so that the user can effectively override the build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now be provided *either* in the environment before running 'make', or as variable assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS is no longer necessary, so they are no longer documented, but are still supported so as not to break existing build systems that supply them when building Asterisk. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207680 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-15Merged revisions 206635 via svnmerge from seanbright1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r206635 | seanbright | 2009-07-15 11:57:51 -0400 (Wed, 15 Jul 2009) | 1 line Only print debug info in codec_dahdi if we are asking for it. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206636 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-18fixes some memory leaks and redundant conditionsdvossel1-1/+1
(closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201678 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-15Shuttle some bits around to address some gain issues with G.722.russell2-6/+6
(closes AST-209) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194722 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-15Further simplify codec_g722 build.russell2-26/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194718 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-15Actually force running make for g722.russell1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194714 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Several changes to codec_dahdi to play nice with G723.sruffell1-84/+325
This commit brings in the changes that were living out on the svn/asterisk/team/sruffell/asterisk-trunk-transcoder branch. codec_dahdi.c now always uses signed linear as the simple codec so that a soft g729 codec will not end up being preferred to the hardware codec. There are also changes to allow codec_dahdi.c to feed packets to the hardware in the native sample size of the codec. This solves problems with choppy audio when using G723. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176760 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-20Merged revisions 157859 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r157859 | kpfleming | 2008-11-19 15:34:47 -0600 (Wed, 19 Nov 2008) | 7 lines the gcc optimizer frequently finds broken code (use of uninitalized variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems. with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course). while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157974 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-30fix a few small things found by using sparsekpfleming1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152809 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-17Merge codec_consistency branch. This should make sample usage much happier.qwell36-1052/+358
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150729 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-15When using MALLOC_DEBUG, codec_lpc10 leaks memory, because it matches a librarytilghman1-1/+1
malloc() with an ast_free (which, of course, doesn't match up with known allocated memory, so the free fails). (closes issue #13702) Reported by: eliel Patches: codec_lpc10_lpcini.c uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@149637 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman9-27/+9
when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-02Update instructions for getting libresamplerussell1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140566 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-20Remove extraneous debugging messages.sruffell1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139154 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-20Fix bug where the samples were not accurate when in G723 mode, which wouldsruffell1-5/+8
cause the timestamp field of the RTP header to be invalid. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@139153 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-09More RSW merges. This should do it for the channels/ dir.seanbright1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136917 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07Updating codec_dahdi to the new transcoder interface.sruffell1-108/+80
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136676 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-07More merges from resolve-shadow warnings:seanbright2-5/+5
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
2008-07-21Remove libresample from the Asterisk source tree. It is now available in itsrussell1-1/+8
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. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132390 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Enable higher quality resampling, as it doesn't have a noticeable performancerussell1-1/+1
impact on my machine .. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132388 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11Janitor patch to change uses of sizeof to ARRAY_LENbbryant3-6/+6
(closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130129 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-26Convert casts to unions, to fix alignment issues on Solaristilghman11-24/+24
(closes issue #12932) Reported by: snuffy Patches: bug_12932_20080627.diff uploaded by snuffy (license 35) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125386 f38db490-d61c-443f-a65b-d21fe96a405b
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