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${ noResults }
234 Commits (a2b47e4edaaf4cb6ce8984745d2f99b4538f2a5b)
Author | SHA1 | Message | Date |
---|---|---|---|
|
a2b47e4eda |
Merged revisions 265320,265467 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265320 | twilson | 2010-05-24 14:06:40 -0500 (Mon, 24 May 2010) | 14 lines Add the FullyBooted AMI event It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ ........ r265467 | twilson | 2010-05-24 17:21:58 -0500 (Mon, 24 May 2010) | 1 line Merge the rest of the FullyBooted patch ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@265521 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
60b18b4ea0 |
Merged revisions 265316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r265316 | tilghman | 2010-05-24 13:19:08 -0500 (Mon, 24 May 2010) | 7 lines 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/branches/1.6.2@265318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
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 |
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Merged revisions 255952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r255952 | tilghman | 2010-04-02 15:19:01 -0500 (Fri, 02 Apr 2010) | 8 lines 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/branches/1.6.2@255955 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 252362 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r252362 | tilghman | 2010-03-14 20:37:04 -0500 (Sun, 14 Mar 2010) | 11 lines 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/branches/1.6.2@252365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 248861 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r248861 | tilghman | 2010-02-25 15:22:39 -0600 (Thu, 25 Feb 2010) | 22 lines 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/branches/1.6.2@248864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 244927 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r244927 | seanbright | 2010-02-05 12:05:32 -0500 (Fri, 05 Feb 2010) | 9 lines 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/branches/1.6.2@244930 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Merged revisions 240499-240500 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r240499 | tilghman | 2010-01-15 15:40:14 -0600 (Fri, 15 Jan 2010) | 9 lines 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 ........ r240500 | tilghman | 2010-01-15 15:42:36 -0600 (Fri, 15 Jan 2010) | 2 lines Oops, missed an include ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@240503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 221777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009) | 9 lines 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/branches/1.6.2@221780 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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AST-2009-005
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Merged revisions 205120 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@205148 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 201678 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r201678 | dvossel | 2009-06-18 11:37:42 -0500 (Thu, 18 Jun 2009) | 11 lines 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/branches/1.6.2@201679 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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 |
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Merged revisions 195320 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r195320 | tilghman | 2009-05-18 14:17:15 -0500 (Mon, 18 May 2009) | 9 lines 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/branches/1.6.2@195359 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 191700 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191700 | seanbright | 2009-05-02 11:45:07 -0400 (Sat, 02 May 2009) | 1 line Update copyright year to 2009 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@191703 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 191367 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191367 | tilghman | 2009-04-30 12:40:58 -0500 (Thu, 30 Apr 2009) | 3 lines Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@191370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 191283 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r191283 | tilghman | 2009-04-30 01:47:13 -0500 (Thu, 30 Apr 2009) | 11 lines 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/branches/1.6.2@191286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 189077 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r189077 | seanbright | 2009-04-17 15:36:38 -0400 (Fri, 17 Apr 2009) | 1 line Fix copy/paste error with 'transmit silence' flag. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@189081 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 187302 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r187302 | tilghman | 2009-04-08 23:59:05 -0500 (Wed, 08 Apr 2009) | 14 lines 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/branches/1.6.2@187303 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184630 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 | russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines Change g_eid to ast_eid_default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@184632 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184339 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@184343 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 184220 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r184220 | eliel | 2009-03-25 10:38:19 -0400 (Wed, 25 Mar 2009) | 19 lines 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/branches/1.6.2@184256 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 182847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ................ r182847 | russell | 2009-03-17 21:28:55 -0500 (Tue, 17 Mar 2009) | 52 lines Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@182947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 178508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Apparently, a void cast doesn't override warn_unused_result.
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16 years ago |
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The 3 possible errors with pipe(2) are all impossible in this situation.
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16 years ago |
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Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init.
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16 years ago |
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add extra check for sysinfo/sysctl
(closes issue #14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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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 |
16 years ago |
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80f91f6ab7 |
Merged revisions 172438 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 | tilghman | 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) | 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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02094f7fb9 |
Merged revisions 169722 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169722 | tilghman | 2009-01-21 15:02:32 -0600 (Wed, 21 Jan 2009) | 8 lines Extra NULLs in the output cause some terminal types to abort in the middle of a color code, causing terminal weirdness. (closes issue #14130) Reported by: coolmig Patches: 20090121__bug14130.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, coolmig ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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fae3ba7421 |
fix assignment in swapmode plug.
Spotted and fix provided by ys (closes issue #14129) Reported by: ys Tested by: ys git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169369 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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ffb3f301af |
Make asterisk compile on non-amd64 versions of OpenBSD.
The HW_PHYSMEM64 is only available in latest OpenBSD and/or amd64 versions of OpenBSD. Use HW_PHYSMEM when HW_PHYSMEM64 is not available. (closes issue #14129) Reported by: ys Patches: 2009011600_physmem64.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak, jtodd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d7eaa9c6aa |
Fix compilation on FreeBSD and OSX
This started as work to fix the 'core show sysinfo' CLI command but while working on it oej pointed out that read_credentials did not compile neither. So while being there, fix that as well. Thanks for all the testing oej! (closes issue #14129) Reported by: ys Tested by: oej, mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168609 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7c6d9c7235 |
Add option to hide console connect messages
(closes issue #14222) Reported by: jamesgolovich Patches: asterisk-hideconnect.diff.txt uploaded by jamesgolovich (license 176) Tested by: otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168585 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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0bad5fd12e |
Don't include swap.h unless we have swapctl
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168479 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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334a89f754 |
When parsing environment variable ASTERISK_PROMPT, make sure to proceed to the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167125 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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53f788c6b5 |
Fix build issues on Linux after sysinfo related changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d2d96b10ac |
introduce 'core show sysinfo' for systems that dont have the Linux-ish sysinfo stuff but do have sysctl.
(closes issue #13433) Reported by: mvanbaak Patches: 2008121300_sysinfosysctl.diff.txt uploaded by mvanbaak (license 7) with two free calls replaced with ast_free based on feedback on reviewboard Review: http://reviewboard.digium.com/r/91/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c8223fc957 |
Merge ast_str_opaque branch (discontinue usage of ast_str internals)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5e034d9f0b |
Merged revisions 163761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163761 | tilghman | 2008-12-12 16:03:10 -0600 (Fri, 12 Dec 2008) | 7 lines Simple fix for Ctrl-C not immediately exiting Asterisk, but also add a pointer inside editline to look back to asterisk.c, so others don't spend as much time as I did looking (in the wrong place) for the appropriate function. Reported by: ZX81, via the #asterisk-users channel Fixed by: me (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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592cab8202 |
Merged revisions 163383 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163383 | tilghman | 2008-12-11 17:35:55 -0600 (Thu, 11 Dec 2008) | 9 lines When a Ctrl-C or Ctrl-D ends a remote console, on certain shells, the terminal is messed up. By intercepting those events with a signal handler in the remote console, we can avoid those issues. (closes issue #13464) Reported by: tzafrir Patches: 20081110__bug13464.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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c8c8995b70 |
add tab completion for 'core set debug X filename.c'
(closes issue #13969) Reported by: jtodd Patches: 20081205__bug13969.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak, eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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d0bc22b3e8 |
Add some additional Asterisk project developer documentation.
After the nightly update of the documentation on asterisk.org, I'll post an update to asterisk-dev with a pointer to the changes. This covers some release branch and commit policy information. None of this should be a surprise, since it's just documenting what we have already been doing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162418 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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179667088b |
Merged revisions 162413 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162413 | russell | 2008-12-09 16:17:39 -0600 (Tue, 09 Dec 2008) | 8 lines Remove the test_for_thread_safety() function completely. The test is not valid. Besides, if we actually suspected that recursive mutexes were not working, we would get a ton of LOG_ERROR messages when DEBUG_THREADS is turned on. (inspired by a discussion on the asterisk-dev list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@162414 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1e8e12efcf |
Janitor, use ARRAY_LEN() when possible.
(closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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033bffd32f |
Introduce CLI permissions.
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: https://origsvn.digium.com/svn/asterisk/trunk@160062 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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e2133648f9 |
Merged revisions 156164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines Move the sanity check that makes sure "always fork" is not set along with the console option to be after the code that reads options from asterisk.conf. This resolves a situation where Asterisk can start taking up 100% when misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to figure out what was causing the 100% CPU problem.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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86f900b201 |
This commit does two things:
- Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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23adb8e509 |
Move all the XML documentation API from pbx.c to xmldoc.c.
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: https://origsvn.digium.com/svn/asterisk/trunk@155711 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |