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2010-09-02Merged revisions 284610 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-20Add load priority order, such that preload becomes unnecessary in most casesTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Oops, merge reverted this fix.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Remove the old stub files, preferring the optional_api method.Tilghman Lesher
(closes issue #17475) Reported by: tilghman Review: https://reviewboard.asterisk.org/r/695/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14Don't try to call an embedded module's backup_globals() function untilKevin P. Fleming
after confirming it exists. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-09Merged revisions 275182 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275182 | mnicholson | 2010-07-09 13:23:23 -0500 (Fri, 09 Jul 2010) | 2 lines give a better error message when attempting to unload a module that is not loaded ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-09Merged revisions 275143 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275143 | mnicholson | 2010-07-09 12:50:05 -0500 (Fri, 09 Jul 2010) | 2 lines don't unload modules that returned AST_MODULE_LOAD_DECLINE when they were loaded ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19Fix transcode_via_sln option with SIP calls and improve PLC usage.Mark Michelson
From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-12Notify CLI when modules is loaded / unloadedPaul Belanger
(closes issue #17308) Reported by: pabelanger Patches: cli.modules.patch uploaded by pabelanger (license 224) Tested by: pabelanger, russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-17RTP documentation states that you can pass NULL as the module, so make sure ↵Tilghman Lesher
that's really the case. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-27Add new option to asterisk.conf (lockconfdir) to protect conf dir during reloadsJeff Peeler
(closes issue #16358) Reported by: raarts Patches: lockconfdir.diff uploaded by raarts (license 937) modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Fixing trunk in a way so that it compiles again.Olle Johansson
Thanks, Philippe :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229871 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-13Add the capability to require a module to be loaded, or else Asterisk exits.Olle Johansson
Review: https://reviewboard.asterisk.org/r/426/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-09Fix various problems detected with Valgrind.Tilghman Lesher
* chan_console accessed pvts after deallocation. * cdr_mysql stored a pointer that was freed by realloc() * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. * The event subsystem sometimes creates an event with no IEs. Due to a corner condition, the code would read beyond the memory boundary. * res_pktccops did not correctly check whether its monitor thread was started. (closes issue #16062) Reported by: alexanderheinz Patches: 20091109__issue16062.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@228798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-21Make LOAD_ORDER actually workTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@213450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-27Fixing typos. Replaces "recieved" with "received" and "initilize" with ↵David Brooks
"initialize" (closes issue #15571) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.Russell Bryant
CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-22attempting to load running modulesDavid Vossel
Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15More 'static' qualifiers on module global variables.Kevin P. Fleming
The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-09module load priorityDavid Vossel
This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-04Merged revisions 199022 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
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: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-02Merge in the RTP engine API.Joshua Colp
This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19Merged revisions 183241 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r183241 | russell | 2009-03-19 12:52:52 -0500 (Thu, 19 Mar 2009) | 2 lines Remove the use of RTLD_NOLOAD, as it is not behaving like expected. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-17Merge a large set of updates to the Asterisk indications API.Russell Bryant
This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingTilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-28remove remaining Zaptel references in various placesKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-18Merged revisions 131921 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul 2008) | 2 lines remove the dlfcn compatibility stuff, because no platforms that Asterisk currently runs on it use it, and it doesn't build anyway ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵Jeff Peeler
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-05Conditionally load the AGI command gosub, depending on whether or not res_agiTilghman Lesher
has been loaded, fix a return value in the loader, and ensure that the help workhorse header does not print on load. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26Add the "config reload <conffile>" command, which allows you to tell AsteriskTilghman Lesher
to reload any file that references a given configuration file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12Revert several changes from revision 102525, as the changes were notTilghman Lesher
compatible, and, in fact, introduced regressions. (Closes issue #12190) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05Create a centralized configuration option for silencethresholdTilghman Lesher
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04Whitespace changes onlyTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-27Merged revisions 104596 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4 lines Use the lock (which already existed, it just wasn't used) on the updaters list to protect the contents instead of the overall module list lock. (closes issue #12080) Reported by: ChaseVenters ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-19Add a log message that appears when you try to unload a module that isn't ↵Joshua Colp
loaded. (closes issue #12033) Reported by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by jamesgolovich (license 176) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-15Merged revisions 103728 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008) | 4 lines In the case that you try to directly reload a module has returned AST_MODULE_LOAD_DECLINE, log a message indicating that the module is not fully initialized and must be initialized using "module load". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-15Merged revisions 103726 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008) | 6 lines Don't attempt to execute the reload callback for a module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash that was reported against chan_console in trunk. (closes issue #11953, reported by junky, fixed by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-05Get rid of any remaining ast_verbose calls in the code in favor of Mark Michelson
ast_verb (closes issue #11934) Reported by: mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-23Move code from res_features into (new file) main/features.cJason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-07Move ModuleLoad and ModuleCheck manager commands from loader.c to manager.c. ↵Joshua Colp
Previously they would get registered twice because of the way manager.c operates. (closes issue #11699) Reported by: caio1982 Patches: manager_module_commands1.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-02some cleanup of this code while I am trying to debug a problem withLuigi Rizzo
gdb dying while debugging asterisk. The problem seems to be related with a race in the handling of module_list, which in turn is triggeded by calling dlopen() on a system which uses initializers to create locks. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06Merged revisions 91366 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4 lines Make sure logger is reloaded at general reload in the cli. (Discovered during Asterisk training in Portugal) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27More "moremanager" fixes. Manager commands to check module status.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21remove a bunch of useless #include "options.h"Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20move asterisk/paths.h outside asterisk.h and into those filesLuigi Rizzo
who really need it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20move internal function declarations to include/asterisk/_private.hLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20switch compile-time option checking to string storage mode in this branch tooKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19another bunch of include removals (errno.h and asterisk/logger.h)Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17define RTLD_LOCAL for platforms that don't have it.Luigi Rizzo
This is only to complete the build, clearly the linker behaviour will be completely different and likely to cause trouble in those cases. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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