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2009-06-15Last batch of 'static' qualifiers for module-level global variables.Kevin P. Fleming
Fix up modules in the 'apps' directory, and also correct the bad example of enum definitions in include/asterisk/app.h, which many developers followed (thanks for reading the documentation!). In addition, add some basic usage examples of the 'pahole' and 'pglobal' tools to the coding guidelines. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15More 'static' qualifiers on module global variables.Kevin P. Fleming
The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15Redesigned 'optional API' support.Kevin P. Fleming
This patch provides a new implementation of the optional API support defined in asterisk/optional_api.h; this new version provides solves compatibility issues with the use of linker version scripts for suppressing global symbols. In addition, there is now a functional (and tested!) implementation for Mac OS/X, so module writers no longer need to use special tests before calling optional API functions. All future implementations must provide these same semantics, so that module writers can rely on them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10Merged revisions 199856 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-09module load priorityDavid Vossel
This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-08Merged revisions 199626,199628 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-05Merged revisions 199297 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-04Merged revisions 199022 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-02Generic call forward api, ast_call_forward()David Vossel
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01Add the ability to execute connected line interception macros.Mark Michelson
When connected line updates are received or generated in the middle of an application call, it is now possible to execute a macro to manipulate the connected line data. This way, phone numbers may be manipulated to be more presentable to users, names may be changed for...whatever reason, or whatever else needs to be done may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-01Minor whitespace fix.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-31Constify the ast_frame arg to ast_queue_frame().Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-29Merged revisions 198068 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Update references to downloads.digium.com to its new URL.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Update references to bugs.digium.com and reviewboard.digium.com to the new URLs.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Add Calendaring support for AsteriskTerry Wilson
This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS Exchange calendars. Exchange support has only been tested on Exchange Server 2k3 and does not support forms-based authentication at this time (patches *very* welcome). Exchange support is also currently missing the ability to return a list of a meting's attendees (again, patches are very, very welcome). Features include: Querying a calendar for events over a specific time range Checking a calendar's busy status via the dialplan Writing calendar events via the dialplan (CalDAV and Exchange only) Handling calendar event notifications through the dialplan (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash Review: https://reviewboard.asterisk.org/r/58 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Merged revisions 197537 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-27Ensure that this header includes xmldoc.h, since it depends on it.Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26Update configure script to check for OSP toolkit 3.5.0.Russell Bryant
(closes issue #14988) Reported by: tzafrir Patches: configure.ac.diff uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick (license 91) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26Add new ast_complete_applications function so that we can use it with theSean Bright
'channel originate ... application <app>' CLI command. (And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop, wanna fight about it!?) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-23Correct example for CLI autocompletion (generation)Kevin P. Fleming
Reported by Atis on #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Implement a new element in AstXML for AMI actions documentation.Eliel C. Sardanons
A new xml element was created to manage the AMI actions documentation, using AstXML. To register a manager action using XML documentation it is now possible using ast_manager_register_xml(). The CLI command 'manager show command' can be used to show the parsed documentation. Example manager xml documentation: <manager name="ami action name" language="en_US"> <synopsis> AMI action synopsis. </synopsis> <syntax> <xi:include xpointer="xpointer(...)" /> <-- for ActionID <parameter name="header1" required="true"> <para>Description</para> </parameter> ... </syntax> <description> <para>AMI action description</para> </description> <see-also> ... </see-also> </manager> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Merged revisions 195881 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Recorded merge of revisions 195366 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-15Add ability for modules to dynamically register logger levelsKevin P. Fleming
This patch adds the ability for modules to dynamically create logger levels for their own use; these are named levels just like the built-in levels, and can be directed to any destination that the logger can send any level to, by including their names in logger.conf. Review: https://reviewboard.asterisk.org/r/244/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-09Minor documentation update for ast_event_queue().Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05Add a more efficient way of allocating structures that use stringfieldsKevin P. Fleming
This commit adds an API call that can be used to allocate a structure along with this stringfield storage in a single allocation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05Correct some flaws in the memory accounting code for stringfields and ao2 ↵Kevin P. Fleming
objects Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192357 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05Properly account for memory allocated for channels and datastoresKevin P. Fleming
As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-05Ensure that string pools allocated to hold stringfields are properly ↵Kevin P. Fleming
accounted in MALLOC_DEBUG mode This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-04Restore 'asyncagi break' command to 1.6.1 and higher.Tilghman Lesher
(closes issue #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt uploaded by tilghman (license 14) 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: nikkk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-04Ensure that astobj2 memory allocations are properly accounted for when ↵Kevin P. Fleming
MALLOC_DEBUG is used This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-03Add 'bitflags'-style information elements to event frameworkKevin P. Fleming
This patch add a new payload type for information elements, a set of bit flags. The payload is transported as a 32-bit unsigned integer but when matching is performed between events and subscribers, the matching is done by using a bitwise AND instead of numeric value comparison. Review: http://reviewboard.asterisk.org/r/242/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-02Remove rarely-used event_log/LOG_EVENT supportKevin P. Fleming
In discussions today at the Europe Asterisk Developer Meet-Up, we determined that the event_log was used in only 9 places in the entire tree, and really was not needed at all. The users have been converted to use LOG_NOTICE, or the messages have been removed since other messages were already in place that provided the same information. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-30Add buffer and echo canceller control to CHANNEL() dialplan function for ↵Kevin P. Fleming
DAHDI channels Adds ability for CHANNEL() dialplan function, when used on DAHDI channels, to temporarily change the number of buffers and/or the buffer policy, and also to enable, disable, or switch the echo canceller between FAX/data and voice modes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-30Detect eaccess (or euidaccess) before using it.Tilghman Lesher
Reported by Andrew Lindh via the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-29SIP option to specify outbound TLS/SSL client protocol.David Vossel
chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip. (closes issue #14770) Reported by: TheOldSaint (closes issue #14768) Reported by: TheOldSaint Review: http://reviewboard.digium.com/r/240/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-29Merge str_substitution branch.Tilghman Lesher
This branch adds additional methods to dialplan functions, whereby the result buffers are now dynamic buffers, which can be expanded to the size of any result. No longer are variable substitutions limited to 4095 bytes of data. In addition, the common case of needing buffers much smaller than that will enable substitution to only take up the amount of memory actually needed. The existing variable substitution routines are still available, but users of those API calls should transition to using the dynamic-buffer APIs. Reviewboard: http://reviewboard.digium.com/r/174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-29Consistent SSL/TLS options across conf filesDavid Vossel
ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though. Review: http://reviewboard.digium.com/r/237/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-29Resolve Solaris build issues and add some API documentation.Russell Bryant
(issue #14981) Reported by: snuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-27Merged revisions 190721 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines Fix 'inconsistent line endings' when autoconf 2.63 is used Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24TLS/SSL private key optionDavid Vossel
Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up. Review: http://reviewboard.digium.com/r/234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24Update comment.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24Add \since tag for new API calls.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24Convert the ast_channel data structure over to the astobj2 framework.Russell Bryant
There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-23Support HTTP digest authentication for the http manager interface.Tilghman Lesher
(closes issue #10961) Reported by: ys Patches: digest_auth_r148468_v5.diff uploaded by ys (license 281) SVN branch http://svn.digium.com/svn/asterisk/team/group/manager_http_auth Tested by: ys, twilson, tilghman Review: http://reviewboard.digium.com/r/223/ Reviewed by: tilghman,russellb,mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-22Merged revisions 190092 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190092 | tilghman | 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) | 7 lines Detect availability of pthread_rwlock_timedwrlock() before using it. (closes issue #14930) Reported by: tilghman Patches: 20090420__bug14930.diff.txt uploaded by tilghman (license 14) Tested by: mvanbaak, tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-22Fix building of chan_h323 with gcc-3.3Jeff Peeler
There seems to be a bug with old versions of g++ that doesn't allow a structure member to use the name list. Rename list member to group_list in ast_group_info and change the few places it is used. (closes issue #14790) Reported by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-21Merged revisions 189601 via svnmerge from Doug Bailey
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189601 | dbailey | 2009-04-21 09:00:55 -0500 (Tue, 21 Apr 2009) | 3 lines Add check in configure script to check for GLOB_NOMAGIC and GLOB_BRACE in glob.h This allows config.c to compile when linked against uclibc that does not support these parameters ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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