asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	1555c082e3	Merged revisions 272925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272925 \| tilghman \| 2010-06-28 16:50:02 -0500 (Mon, 28 Jun 2010) \| 8 lines Don't change ownership/group/permissions on run directory, if it already exists. (closes issue #17076) Reported by: stuarth Patches: 20100324__issue17076.diff.txt uploaded by tilghman (license 14) Tested by: stuarth ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5d313f51b9	Merged revisions 269635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r269635 \| tilghman \| 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010) \| 9 lines Ensure restartable system calls can restart (BSD signal semantics). This eliminates the annoying <beep> on the console. (closes issue #17477) Reported by: jvandal Patches: 20100610__issue17477.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269636 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	afd4454c44	Generic Advice of Charge. Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	dd26c53707	Merged revisions 266585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266585 \| tilghman \| 2010-06-01 10:17:46 -0500 (Tue, 01 Jun 2010) \| 11 lines Prevent CLI prompt from distorting output of lines shorter than the prompt. Uses the VT100 method of clearing the line from the cursor position to the end of the line: Esc-0K (closes issue #17160) Reported by: coolmig Patches: 20100531__issue17160.diff.txt uploaded by tilghman (license 14) Tested by: coolmig ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2da88f1977	Setup environment variables for the benefit of child processes and disallow changing them. (closes issue #14899) Reported by: jmls Patches: 20090916__issue14899.diff.txt uploaded by tilghman (license 14) Tested by: jmls git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	7e204048fc	Only report swap on platforms which can examine those statistics git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	fb80119b87	Merged revisions 266142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266142 \| tilghman \| 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) \| 14 lines Use sigaction for signals which should persist past the initial trigger, not signal. If you call signal() in a Solaris signal handler, instead of just resetting the signal handler, it causes the signal to refire, because the signal is not marked as handled prior to the signal handler being called. This effectively causes Solaris to immediately exceed the threadstack in recursive signal handlers and crash. (closes issue #17000) Reported by: rmcgilvr Patches: 20100526__issue17000.diff.txt uploaded by tilghman (license 14) Tested by: rmcgilvr ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	0390dae08d	Merge the rest of the FullyBooted patch git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	6f998f06af	On systems with a LOT of RAM, a signed integer sometimes printed negative. (closes issue #16837) Reported by: jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6bb45831eb	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/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	a753e8878b	Asterisk data retrieval API. This module implements an abstraction for retrieving and exporting asterisk data. Developed by: Brett Bryant <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY) <eliels@gmail.com> For the Google Summer of code 2009 Project. Documentation can be found in doxygen format and inside the header include/asterisk/data.h Review: https://reviewboard.asterisk.org/r/275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3c6c879681	Pass the PID of the Asterisk process, not the PID of the canary. (closes issue #17065) Reported by: globalnetinc Patches: astcanary.patch uploaded by makoto (license 38) Tested by: frawd, globalnetinc git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	5a5599a764	make 'core show settings' should show all settable directories (closes issue #17086) Reported by: tzafrir Patches: asterisk_extra_settings_dirs.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	f9e4d024c9	Initialize channels prior to loading "preload" modules. We can have bad results when a module, upon being loaded, attempts to reference the channels container if the container hasn't yet been initialized. I saw this happen by trying to preload pbx_config.so and having a hint defined which referenced a non-existent SIP peer. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253872 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	32a649c3b9	Update comment to reflect new timeout value. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	d6ca5ac86e	Increase CLI command output timeout for asterisk -rx to 60 seconds. (closes issue #17049) Reported by: russell Tested by: russell Review: https://reviewboard.asterisk.org/r/573/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	4c8f14b7cf	Merged revisions 252361 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r252361 \| tilghman \| 2010-03-14 20:33:50 -0500 (Sun, 14 Mar 2010) \| 4 lines Launch Asterisk on Mac OS X with launchd. Reviewboard: https://reviewboard.asterisk.org/r/551/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a0af2cff0e	Merged revisions 248859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r248859 \| tilghman \| 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) \| 15 lines Some platforms clear /var/run at boot, which makes connecting a remote console... difficult. Previously, we only created the default /var/run/asterisk directory at install time. While we could create it in the init script, that would not work for those who start asterisk manually from the command line. So the safest thing to do is to create it as part of the Asterisk boot process. This also changes the ownership of the directory, because the pid and ctl files are created after we setuid/setgid. (closes issue #16802) Reported by: Brian Patches: 20100224__issue16802.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	acdb1bd82f	Merged revisions 244926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r244926 \| seanbright \| 2010-02-05 12:03:35 -0500 (Fri, 05 Feb 2010) \| 1 line Update main copyright date. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	a170cd28e0	Add new option to asterisk.conf (lockconfdir) to protect conf dir during reloads (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	16 years ago
Tilghman Lesher	6c1e69370e	Err, oops, it was already the way I intended. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8037982fa6	The previous attempt at using a pipe to guarantee astcanary shutdown did not work. We're revisiting the previous patch, albeit with a method that overcomes the prior criticism that it was not POSIX-compliant. (closes issue #16602) Reported by: frawd Patches: 20100114__issue16602.diff.txt uploaded by tilghman (license 14) Tested by: frawd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	5e0762370c	It's been long enough -- make the behavior introduced in 1.6 the default. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@239000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	73cb2d507b	Unit Test Framework API The Unit Test Framework is a new API that manages registration and execution of unit tests in Asterisk with the purpose of verifying the operation of C functions. The Framework consists of a single test manager accompanied by a list of registered test functions defined within the code. A test is defined, registered, and unregistered from the framework using a set of macros which allow the test code to only be compiled within asterisk when the TEST_FRAMEWORK flag is enabled in menuselect. This allows the test code to exist in the same file as the C functions it intends to verify. Registered tests may be viewed and executed via a set of new CLI commands. CLI commands are also present for generating and exporting test results into xml and txt formats. For more information and use cases please refer to the documentation provided at the beginning of the test.h file. Review: https://reviewboard.asterisk.org/r/447/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Joshua Colp	d0e431ce3d	Add an 'X' option to the asterisk application which enables #exec for configuration files. This option can be used to enable #exec support in the asterisk.conf configuration file. (closes issue #16260) Reported by: atis Patches: exec_includes.patch uploaded by atis (license 242) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232510 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	23b4a700f4	Reorder option flags. Change guidelines so that example code is consistent with guidelines git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	75c015bfff	Add the capability to require a module to be loaded, or else Asterisk exits. Review: https://reviewboard.asterisk.org/r/426/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	844a01b27e	Add an "Asterisk Architecture Overview" section to the doxygen documentation. This is a side project I've been poking at this week. The intent is to discuss Asterisk architecture in a top down fashion to help new developers understand how Asterisk is put together. There is a ton of stuff to write about, so this will just continue to evolve over time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	8c7b3cf738	Merged revisions 221776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221776 \| tilghman \| 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) \| 2 lines Fix a bunch of off-by-one errors ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@221777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	17180120bf	Change the default behavior of Set, AGI, and pbx_realtime to 1.6 behavior by default (starting in 1.6.3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	642bec4d6f	AST-2009-005 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0e8c630224	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: https://origsvn.digium.com/svn/asterisk/trunk@205120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. 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	17 years ago
David Vossel	dcfe69ec64	fixes some memory leaks and redundant conditions (closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	befad10893	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/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3982b8c71b	Call ast_stun_init() when we're initializing to get the 'stun debug set' commands. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) 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	17 years ago
Tilghman Lesher	92ace48782	Move the spawn of astcanary down, until after the call to daemon(3). This avoids possible conflicts with the internal implementation of daemon(3). (closes issue #15093) Reported by: tzafrir Patches: 20090513__issue15093__2.diff.txt uploaded by tilghman (license 14) Tested by: tzafrir git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f2d26b3ddb	Update copyright year to 2009 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	91dde03ba8	Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4cbe6b1afe	Change working directory to / under certain conditions. If backgrounding and no core will be produced, then changing the directory won't break anything; likewise, if the CWD isn't accessible by the current user, then a core wasn't possible anyway. (closes issue #14831) Reported by: chris-mac Patches: 20090428__bug14831.diff.txt uploaded by tilghman (license 14) 20090430__bug14831.diff.txt uploaded by tilghman (license 14) Tested by: chris-mac git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	7a7f17ce4d	Fix copy/paste error with 'transmit silence' flag. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	8f28bfc63e	Merged revisions 187300-187301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187300 \| tilghman \| 2009-04-08 23:31:38 -0500 (Wed, 08 Apr 2009) \| 3 lines Add debugging mode for diagnosing file descriptor leaks. (Related to issue #14625) ........ r187301 \| tilghman \| 2009-04-08 23:32:40 -0500 (Wed, 08 Apr 2009) \| 2 lines Oops, missed this file in the last commit. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2cb0018fa1	Start splitting up miscellaneous doxygen documentation into separate files. doxyref.h was created to hold miscellaneous documentation that was not specific to a part of the code. This file has grown quite a bit so I decided to start splitting parts of it out into new files. Now, you can drop a new file into include/asterisk/doxygen/ and it will be processed by doxygen. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	63de834395	Merge in the RTP engine API. 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	17 years ago
Russell Bryant	b564b2105f	Change g_eid to ast_eid_default. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ee77b475f2	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: https://origsvn.digium.com/svn/asterisk/trunk@184339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	79ec90b3b2	Merged revisions 184188 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184188 \| eliel \| 2009-03-25 10:12:54 -0400 (Wed, 25 Mar 2009) \| 13 lines Avoid destroying the CLI line when moving the cursor backward and trying to autocomplete. When moving the cursor backward and pressing TAB to autocomplete, a NULL is put in the line and we are loosing what we have already wrote after the actual cursor position. (closes issue #14373) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0bdd99ad64	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: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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