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2009-07-08Merged revisions 205120 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@205148 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-04Merged revisions 199051 via svnmerge from seanbright1-0/+12
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: http://svn.digium.com/svn/asterisk/branches/1.6.2@199054 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-25Merged revisions 184339 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r184339 | russell | 2009-03-25 16:57:19 -0500 (Wed, 25 Mar 2009) | 35 lines Improve performance of the ast_event cache functionality. This code comes from svn/asterisk/team/russell/event_performance/. Here is a summary of the changes that have been made, in order of both invasiveness and performance impact, from smallest to largest. 1) Asterisk 1.6.1 introduces some additional logic to be able to handle distributed device state. This functionality comes at a cost. One relatively minor change in this patch is that the extra processing required for distributed device state is now completely bypassed if it's not needed. 2) One of the things that I noticed when profiling this code was that a _lot_ of time was spent doing string comparisons. I changed the way strings are represented in an event to include a hash value at the front. So, before doing a string comparison, we do an integer comparison on the hash. 3) Finally, the code that handles the event cache has been re-written. I tried to do this in a such a way that it had minimal impact on the API. I did have to change one API call, though - ast_event_queue_and_cache(). However, the way it works now is nicer, IMO. Each type of event that can be cached (MWI, device state) has its own hash table and rules for hashing and comparing objects. This by far made the biggest impact on performance. For additional details regarding this code and how it was tested, please see the review request. (closes issue #14738) Reported by: russell Review: http://reviewboard.digium.com/r/205/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@184343 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merge a large set of updates to the Asterisk indications API.russell1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-01Introduce CLI permissions.eliel1-0/+1
Based on cli_permissions.conf configuration file, we are able to permit or deny cli commands based on some patterns and the local user and group running rasterisk. (Sorry if I missed some of the testers). Reviewboard: http://reviewboard.digium.com/r/11/ (closes issue #11123) Reported by: eliel Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak git-svn-id: http://svn.digium.com/svn/asterisk/trunk@160062 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-10Move all the XML documentation API from pbx.c to xmldoc.c.eliel1-2/+2
Export the XML documentation API: ast_xmldoc_build_synopsis() ast_xmldoc_build_syntax() ast_xmldoc_build_description() ast_xmldoc_build_seealso() ast_xmldoc_build_arguments() ast_xmldoc_printable() ast_xmldoc_load_documentation() git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155711 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01Merge changes from team/group/appdocsxmlrussell1-0/+6
This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-16Add a "timing test" CLI command. It opens a timer and configures it forrussell1-0/+1
50 ticks per second, and then counts to see how many ticks it actually gets in a second. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122926 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-03A taskprocessor is an object that has a name, a task queue, and an event ↵dhubbard1-0/+1
processing thread. Modules reference a taskprocessor, push tasks into the taskprocessor as needed, and unreference the taskprocessor when the taskprocessor is no longer needed. A task wraps a callback function pointer and a data pointer and is managed internal to the taskprocessor subsystem. The callback function is responsible for releasing task data. Taskprocessor API * ast_taskprocessor_get(..) - returns a reference to a taskprocessor * ast_taskprocessor_unreference(..) - releases reference to a taskprocessor * ast_taskprocessor_push(..) - push a task into a taskprocessor queue Check doxygen for more details git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115268 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-18start the process of changing HTTP request dispatching to do it based on ↵kpfleming1-1/+3
*both* URI and method, so that POST support can move into a module; move http.c's private function prototypes into _private.h git-svn-id: http://svn.digium.com/svn/asterisk/trunk@109762 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-033) In addition to merging the changes below, change trunk back to a regularrussell1-0/+1
LIST instead of an RWLIST. The way this list works makes it such that a RWLIST provides no additional benefit. Also, a mutex is needed for use with the thread condition. Merged revisions 105563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines Merge in some changes from team/russell/autoservice-nochans-1.4 These changes fix up some dubious code that I came across while auditing what happens in the autoservice thread when there are no channels currently in autoservice. 1) Change it so that autoservice thread doesn't keep looping around calling ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition so that the thread properly goes to sleep and does not wake up until a channel is put into autoservice. This actually fixes an interesting bug, as well. If the autoservice thread is already running (almost always is the case), then when the thread goes from having 0 channels to have 1 channel to autoservice, that channel would have to wait for up to 1/2 of a second to have the first frame read from it. 2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no channels and no fds to poll() on, such as was the case with the previous code for the autoservice thread. In this case, the code would call alloca(0), and pass the result as the first argument to poll(). In this case, the 2nd argument to poll() specified that there were no fds, so this invalid pointer shouldn't actually get dereferenced, but, this code makes it explicit and ensures the pointers are NULL unless we have valid data to put there. (related to issue #12116) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105564 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-23Move code from res_features into (new file) main/features.cqwell1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100039 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-11A lot of doxygen updatesoej1-0/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92285 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22implement the split of file.h and mod_format.hrizzo1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89515 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20move internal function declarations to include/asterisk/_private.hrizzo1-0/+44
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89465 f38db490-d61c-443f-a65b-d21fe96a405b