81 Commits (cef5e9d4340f94de8801aaecfb004a080df8a68f)

Author	SHA1	Message	Date
Kevin P. Fleming	77ad0e2e02	Merged revisions 276441 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r276441 | kpfleming | 2010-07-14 15:15:48 -0500 (Wed, 14 Jul 2010) | 4 lines Don't try to call an embedded module's backup_globals() function until after confirming it exists. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@276442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	f6b650cfb8	Merged revisions 275186 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r275186 | mnicholson | 2010-07-09 13:24:03 -0500 (Fri, 09 Jul 2010) | 9 lines Merged revisions 275182 via svnmerge from 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/branches/1.6.2@275191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	6d677066be	Merged revisions 275144 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r275144 | mnicholson | 2010-07-09 12:50:45 -0500 (Fri, 09 Jul 2010) | 9 lines Merged revisions 275143 via svnmerge from 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/branches/1.6.2@275145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	af727635c5	Merged revisions 264452 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r264452 | mmichelson | 2010-05-19 16:29:08 -0500 (Wed, 19 May 2010) | 86 lines Fix transcode_via_sln option with SIP calls and improve PLC usage. 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/branches/1.6.2@264453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	4855426e56	Merged revisions 262800 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r262800 | pabelanger | 2010-05-12 15:59:16 -0400 (Wed, 12 May 2010) | 8 lines Notify CLI when modules is loaded / unloaded (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/branches/1.6.2@262801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	f4be17e388	Merged revisions 228798 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r228798 | tilghman | 2009-11-09 01:37:52 -0600 (Mon, 09 Nov 2009) | 14 lines Fix various problems detected with Valgrind. * chan_console accessed pvts after deallocation. * The module loader did not check usecount on shutdown, which led to chan_iax2 reading a timer that was already unloaded. (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/branches/1.6.2@248011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	799d82c73e	Merged revisions 213450 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r213450 | twilson | 2009-08-20 22:48:54 -0500 (Thu, 20 Aug 2009) | 2 lines Make LOAD_ORDER actually work ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@213453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	4921462e88	Merged revisions 209098 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@209234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	824737f6d7	Merged revisions 202410 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r202410 | dvossel | 2009-06-22 10:33:35 -0500 (Mon, 22 Jun 2009) | 5 lines attempting to load running modules 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/branches/1.6.2@202411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	443cb28322	Merged revisions 199743 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines module load priority 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/branches/1.6.2@199744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b4b61e5273	Merged revisions 199051 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from 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/branches/1.6.2@199054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	89e2c98dfe	Merged revisions 183242 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r183242 | russell | 2009-03-19 13:00:15 -0500 (Thu, 19 Mar 2009) | 10 lines Merged revisions 183241 via svnmerge from 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/branches/1.6.2@183263 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. ... 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	17 years ago
Tilghman Lesher	08af5bb312	Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating ... 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	18 years ago
Kevin P. Fleming	6291cd19bf	remove remaining Zaptel references in various places ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	9a08061ea3	Merged revisions 131921 via svnmerge from ... 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	18 years ago
Jeff Peeler	ef3b214728	Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. 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	18 years ago
Tilghman Lesher	90d75af346	Conditionally load the AGI command gosub, depending on whether or not res_agi ... 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	18 years ago
Tilghman Lesher	ef4eff9a9b	Add the "config reload <conffile>" command, which allows you to tell Asterisk ... 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	18 years ago
Tilghman Lesher	10609251f9	Revert several changes from revision 102525, as the changes were not ... 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	18 years ago
Tilghman Lesher	8a411ccf83	Create a centralized configuration option for silencethreshold ... (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	18 years ago
Tilghman Lesher	cfc1df4c1a	Whitespace changes only ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	b868d44342	Merged revisions 104596 via svnmerge from ... 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	18 years ago
Joshua Colp	ee5e39e04b	Add a log message that appears when you try to unload a module that isn't 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	18 years ago
Russell Bryant	75fd299f1b	Merged revisions 103728 via svnmerge from ... 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	18 years ago
Russell Bryant	71646d663b	Merged revisions 103726 via svnmerge from ... 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	18 years ago
Mark Michelson	fe9821cc10	Get rid of any remaining ast_verbose calls in the code in favor of ... 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	18 years ago
Jason Parker	3bd33214b9	Move code from res_features into (new file) main/features.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	96f5a494cf	Move ModuleLoad and ModuleCheck manager commands from loader.c to manager.c. 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	18 years ago
Luigi Rizzo	b9aeecdb66	some cleanup of this code while I am trying to debug a problem with ... 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	18 years ago
Olle Johansson	9dc965eadd	Merged revisions 91366 via svnmerge from ... 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	18 years ago
Olle Johansson	a30972ee65	More "moremanager" fixes. Manager commands to check module status. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	e0ff5fef5c	remove a bunch of useless #include "options.h" ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	a23c055c3d	move asterisk/paths.h outside asterisk.h and into those files ... who really need it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	915b97d300	move internal function declarations to include/asterisk/_private.h ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	4d522da4aa	switch compile-time option checking to string storage mode in this branch too ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	9335ace850	another bunch of include removals (errno.h and asterisk/logger.h) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	6cca49b9d0	define RTLD_LOCAL for platforms that don't have it. ... 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	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect ... build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	547306835e	Merged revisions 89325 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007) | 4 lines To help combat problems where people build external modules (asterisk-addons or others) and then change the build options of the Asterisk build in a way that makes the incompatible without warning, this commit introduces an MD5 signature of the important build-time options and includes that signature into modules when they are built. When the loader loads one of these modules and notices the problem, it will emit a warning to console and refuse to initialize the module, as doing so could cause the system to be unstable or even crash. If you upgrade to this version of Asterisk, you must rebuild *all* of your modules that came from other sources before trying to run this version. If you are using Digium's G.729 binary codec module, you will need v33 or newer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89326 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	edc78d6023	improve linked-list macros in two ways: ... - the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	d0b2d2dc62	remove a debugging message which i forgot in. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	81456f8493	Fix embedding of modules on FreeBSD: ... the constructor for the list of modules was run after the constructors for the embedded modules (which appended entries to the list). As a result, the list appeared empty when it was time to use it. On linux the order of execution of constructor was evidently different (it may depend on the ordering of modules in the ELF file). This is only a workaround - there may be other situations where the execution of constructors causes problems, so if we manage to find a more general solution this workaround can go away. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Kevin P. Fleming	6a0ae0aca4	merging patches that don't compile is bad... mmkay? ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	e4b75ab0e0	Merged revisions 86630 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86630 | file | 2007-10-22 10:33:23 -0300 (Mon, 22 Oct 2007) | 6 lines Fixes for building under OpenSolaris. (closes issue #11047) Reported by: snuffy Patches: 11047-fixes.diff uploaded by snuffy (license 35) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	56b9568164	Don't reload a configuration file if nothing has changed. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	f8483a0d04	Do a massive conversion for using the ast_verb() macro ... (closes issue #10277, patches by mvanbaak) Basically, this changes ... if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3, "Something\n"); git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	0e969271ae	After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	81bc1d7af5	Merge in ast_strftime branch, which changes timestamps to be accurate to the microsecond, instead of only to the second ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	8a7732f067	via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago