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2010-03-02Merged revisions 249893 via svnmerge from dvossel1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r249893 | dvossel | 2010-03-02 13:08:38 -0600 (Tue, 02 Mar 2010) | 11 lines fixes adaptive jitterbuffer configuration When configuring the adaptive jitterbuffer, the target_extra value not only could not be set from the configuration, but was not even being set to its proper default. This value is required in order for the adaptive jitterbuffer to work correctly. To resolve this a config option has been added to expose this value to the conf files, and a default value is provided when no config specific value is present. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@249907 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 209400 via svnmerge from tilghman1-3/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238494 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Merged revisions 196988 via svnmerge from seanbright1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines Display an error message when chan_alsa fails to load due to a missing or inaccessible configuration file. Before this change, when chan_alsa failed to load due to a missing or inaccessible configuration file, no message would be displayed. With this change, when chan_alsa fails to load due to a missing or inaccessible configuration file, a message will be displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: chan_alsa.c-confload.patch uploaded by Nick (license 657) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@196989 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-18Merged revisions 182847 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@182945 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-12Merged revisions 168526 via svnmerge from tilghman1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r168526 | tilghman | 2009-01-12 17:45:51 -0600 (Mon, 12 Jan 2009) | 12 lines Merged revisions 167095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167095 | tilghman | 2008-12-31 18:01:22 -0600 (Wed, 31 Dec 2008) | 5 lines Repeat attempts to write when we receive -EAGAIN from the driver, as detailed in the ALSA sample code (see http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32) Reported by: Jerry Geis (via the -users list) Fixed by: me (license 14) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@168527 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-29Merged revisions 118955,118957 via svnmerge from tilghman1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines Add some debugging code that ensures that when we do deadlock avoidance, we don't lose the information about how a lock was originally acquired. ........ ................ r118957 | tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10 lines Merged revisions 118954 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008) | 2 lines Define also when not DEBUG_THREADS ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@118958 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-13Merged revisions 115945 via svnmerge from file1-2/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r115945 | file | 2008-05-13 17:29:27 -0300 (Tue, 13 May 2008) | 12 lines Merged revisions 115944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 lines Use the right flag to open the audio in non-blocking. (closes issue #12616) Reported by: nicklewisdigiumuser ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@115946 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-06Merged revisions 106239 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) | 12 lines Merged revisions 106235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106318 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-03coding guidelines cleanupkpfleming1-312/+51
remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96245 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-2/+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 affectrizzo1-4/+0
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
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵qwell1-5/+5
didn't make much sense git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Convert NEW_CLI to AST_CLI.qwell1-5/+5
Closes issue #11039, as suggested by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-27Merged revisions 83974 via svnmerge from kpfleming1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r83974 | kpfleming | 2007-09-26 16:53:03 -0700 (Wed, 26 Sep 2007) | 2 lines avoid the weird usage of assert() in the ALSA header files that gcc 4.2 wants to complain about ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83986 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-18(issue #10724)qwell1-92/+117
Reported by: eliel Patches: res_features.c.patch uploaded by eliel (license 64) res_agi.c.patch uploaded by seanbright (license 71) res_musiconhold.c.patch uploaded by seanbright (license 71) pbx.c.patch uploaded by moy (license 222) logger.c.patch uploaded by moy (license 222) frame.c.patch uploaded by moy (license 222) manager.c.patch uploaded by moy (license 222) http.c.patch uploaded by moy (license 222) dnsmgr.c.patch uploaded by moy (license 222) res_realtime.c.patch uploaded by eliel (license 64) res_odbc.c.patch uploaded by seanbright (license 71) res_jabber.c.patch uploaded by eliel (license 64) chan_local.c.patch uploaded by eliel (license 64) chan_agent.c.patch uploaded by eliel (license 64) chan_alsa.c.patch uploaded by eliel (license 64) chan_features.c.patch uploaded by eliel (license 64) chan_sip.c.patch uploaded by eliel (license 64) RollUp.1.patch (includes all of the above patches) uploaded by seanbright (license 71) Convert many CLI commands to the NEW_CLI format. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82930 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-17convert various places that access the channel lock directly to use the ↵russell1-8/+6
channel lock wrappers git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82728 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Don't reload a configuration file if nothing has changed.tilghman1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Add support for using epoll instead of poll. This should increase ↵file1-1/+1
scalability and is done in such a way that we should be able to add support for other poll() replacements. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78683 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-26Do a massive conversion for using the ast_verb() macrorussell1-4/+2
(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
2007-06-14Add a massive set of changes for converting to use the ast_debug() macro.russell1-12/+6
(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
2007-05-14Merged revisions 64306 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | 3 lines Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication will trigger an error and cause sounds to stop, which in this case, is ringing. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64322 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-10Merged revisions 60989 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. This also adds the mods from 1.4/r.61136; ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61152 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-26Merged revisions 56888 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) | 4 lines Restore the behavior of Asterisk 1.2 where if a device was not specified in alsa.conf, then we just use the system default, instead of creating our own default of hw:0,0. (issue #9139) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56889 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-23Merged revisions 51788 via svnmerge from file1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51801 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19Merged revisions 51311 via svnmerge from russell1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51314 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-06Constify a bunch of usage strings for CLI commands.russell1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48306 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-07A fair number of changes for the sake of bug 7506murf1-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47290 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-03remove useless usecnt stuffrizzo1-9/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47075 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-25Merged revisions 46200 via svnmerge from kpfleming1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006) | 2 lines apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@46201 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-04Merged revisions 44378 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines update thread creation code a bit reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44379 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-03bug #8076 check option_debug before printing to debug channel.mogorman1-10/+20
patch provided in bugnote, with minor changes. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44253 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-21Clean up chan_alsa load module function (issue #8000 reported by Mithraen)file1-40/+40
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43459 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-21Lots more removal of deprecated thingstilghman1-213/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43452 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18merge qwell's CLI verbification workkpfleming1-24/+246
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43212 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-08Formatting fixes for chan_alsa (issue #7807 reported by Mithraen with more ↵file1-253/+247
mods done by myself) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@42388 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-31Merge in VLDTMF support with Zaptel/Core done by the ever great Darumkilla ↵file1-1/+1
Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41507 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21merge new_loader_completion branch, including (at least):kpfleming1-14/+3
- restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-16move the calls to ast_jb_configure() to before the PBX thread is started on therussell1-2/+1
channel to remove the theoretical race condition that the channel could get bridged before the channel's jitterbuffer gets configured. This was pointed out by PCadach on IRC. Thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@39964 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-19merge Russell's 'hold_handling' branch, finally implementing music-on-hold ↵kpfleming1-13/+28
handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37988 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-23revert my changes that converted the jb on the channel to be dynamicallyrussell1-11/+7
allocated. These changes caused crashes when using a channel type that did not support the jitterbuffer. Instead of fixing why it's crashing, I'm going to implement this in a better way next week. The way I did it caused a jitterbuffer to be allocated on every channel where the channel type supported jitterbuffers, even if they were disabled. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35746 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-22- dynamically allocate the ast_jb structure that is on the channel structurerussell1-7/+11
so that channels not using a jitterbuffer don't waste as much memory - ensure that the channel drivers that use jitterbuffers can handle a failure from configuring a jitterbuffer on a new channel because of a memory allocation error - On passing through these channel drivers, configure the jitterbuffer before starting the PBX thread instead of afterwards. If the pbx fails to start for whatever reason, this would have caused a crash. - Also on passing, move the increase of the usecount to after all of the possible failure conditions in the function - fix a place where ast_update_use_count() was not called - ensure that the owner channel pointer of the channel pvt strcutures is set to NULL in failure conditions git-svn-id: http://svn.digium.com/svn/asterisk/trunk@35553 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-4/+4
again :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-31move the includes of abstract_jb.h to be with the rest of the asterisk includes.russell1-2/+2
These used to be wrapped in a #ifdef git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31078 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-31Add support for using a jitterbuffer for RTP on bridged calls. This includesrussell1-0/+22
a new implementation of a fixed size jitterbuffer, as well as support for the existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov) Thank you very much to Slav Klenov of Securax and all of the people involved in the testing of this feature for all of your hard work! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@31052 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10ensure that control frames with payload can be sent to channel drivers via ↵kpfleming1-2/+2
->indicate() update iax2_indicate to pass control frame payload to the connected channel add an API call for sending an indication with payload, and use it for control frames with payload git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26417 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-24Thanks to the fine work of Russell Bryant and Dancho Lazarov, we now have ↵kpfleming1-11/+9
autoconf and menuselect tools for Asterisk! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@22267 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-17more module loader related fixeskpfleming1-5/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20963 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-08since the module API is changing, it's a good time to const-ify the ↵kpfleming1-2/+2
description() and key() return values git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18552 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-01use string fields for some stuff in ast_channelkpfleming1-10/+9
const-ify some more APIs remove 'type' field from ast_channel, in favor of the one in the channel's tech structure allow string field module users to specify the 'chunk size' for pool allocations update chan_alsa to be compatible with recent const-ification patches git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9060 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-16 03:48:29 +0000