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${ noResults }
69 Commits (master)
Author | SHA1 | Message | Date |
---|---|---|---|
|
82bef2ed90 |
docs: Indent <since> tags.
Also updates the 'since' of applications/functions that existed before XML documentation was introduced (1.6.2.0). |
3 months ago |
|
198b01c536 |
docs: Add version information to application and function XML elements
* Do a git blame on the embedded XML application or function element. * From the commit hash, grab the summary line. * Do a git log --grep <summary> to find the cherry-pick commits in all branches that match. * Do a git patch-id to ensure the commits are all related and didn't get a false match on the summary. * Do a git tag --contains <commit> to find the tags that contain each commit. * Weed out all tags not ..0. * Sort and discard any .0.0 and following tags where the commit appeared in an earlier branch. * The result is a single tag for each branch where the application or function was defined. The applications and functions defined in the following files were done by hand because the XML was extracted from the C source file relatively recently. * channels/pjsip/dialplan_functions_doc.xml * main/logger_doc.xml * main/manager_doc.xml * res/res_geolocation/geoloc_doc.xml * res/res_stir_shaken/stir_shaken_doc.xml |
3 months ago |
|
468601fd13 |
manager: Add `<since>` tags for all AMI actions.
|
3 months ago |
|
cec0c5cd39 |
app_senddtmf: Add SendFlash AMI action.
Adds an AMI action to send a flash event on a channel. ASTERISK-30440 #close Change-Id: I4707aeecb3cd8f3e63fd0c3fe009798943c369c9 |
2 years ago |
|
cee3564b45 |
app_senddtmf: Add option to answer target channel.
Adds a new option to SendDTMF() which will answer the specified channel if it is not already up. If no channel is specified, the current channel will be answered instead. ASTERISK-30422 Change-Id: Iddcbd501fcdf9fef0f453b7a8115a90b11f1d085 |
2 years ago |
|
772b59034f |
app_senddtmf: Add receive mode to AMI Action PlayDTMF
ASTERISK-28614 Change-Id: I183501297ae1dc294ae56b34acac9b0343eb2664 |
6 years ago |
|
7d874c1af7 |
AMI PlayDTMF Action: Make not compete with channel's media thread.
There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905 |
7 years ago |
|
fd0ca1c3f9 |
Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0 |
7 years ago |
|
a6e5bae3ef |
Remove ASTERISK_REGISTER_FILE.
ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966 |
9 years ago |
|
4a58261694 |
git migration: Refactor the ASTERISK_FILE_VERSION macro
Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e |
10 years ago |
|
b5138fccf4 |
Add pause one second W dial modifier.
* The following dialplan applications now recognize 'W' to pause sending DTMF for one second in addition to the previously existing 'w' that paused sending DTMF for half a second. Dial, ExternalIVR, and SendDTMF. * The chan_dahdi analog port dialing and deferred DTMF dialing for PRI now distinguishes between 'w' and 'W'. The 'w' pauses dialing for half a second. The 'W' pauses dialing for one second. * Created dahdi_dial_str() in chan_dahdi that eliminated a lot of duplicated dialing code and diagnostic messages for the channel driver. (closes issue ASTERISK-20039) Reported by: Jeremiah Gowdy Patches: jgowdy-wait-6-22-2012.diff (license #5621) patch uploaded by Jeremiah Gowdy Expanded patch to add support in chan_dahdi. Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
acb3a5f76f |
Add Duration header for PlayDTMF AMI Action
This patch adds an optional header to the PlayDTMF AMI action, Duration. It allows the duration of the DTMF digit to be played on the channel to be specified in milliseconds. (closes issue ASTERISK-18172) Reported by: Renato dos Santos patches: send-dtmf.patch uploaded by Renato dos Santos (license #6267) Modified slightly for this commit for Asterisk 12. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373979 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
02ed1bd638 |
Fix SendDTMF crash and channel reference leak using channel name parameter.
The SendDTMF channel name parameter has two issues. 1) Crashes if the channel name does not exist. 2) Leaks a channel reference if the channel is the current channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF documentation. * Renamed app to senddtmf_name and tweaked the type. ........ Merged revisions 373945 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373946 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373954 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
a525edea59 |
Merged revisions 328247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
cba378d847 |
Allow SendDTMF to play digits to a specified channel.
Patch supplied by reporter was modified to use autoservice and prevent a potential channel ref leak but is otherwise as the reporter uploaded it. (closes issue #17182) Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded by rcasas (license 641) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
0375f18c9f |
Code clean up in app_senddtmf
Pushes code clean up done in app_externalivr back into app_senddtmf Review: https://reviewboard.asterisk.org/r/473/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243346 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
2c882626a0 |
Implement a new element in AstXML for AMI actions documentation.
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 |
16 years ago |
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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 |
16 years ago |
|
cba19c8a67 |
Convert the ast_channel data structure over to the astobj2 framework.
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 |
16 years ago |
|
a7028f2bc6 |
Merged revisions 184078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184078 | mmichelson | 2009-03-24 17:34:45 -0500 (Tue, 24 Mar 2009) | 9 lines Change NULL pointer check to be ast_strlen_zero. The 'digit' variable is guaranteed to be non-NULL, so the if statement could never evaluate true. Changing to ast_strlen_zero makes the logic correct. This was found while reviewing ast_channel_ao2 code review. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184079 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
d23dff9ca8 |
- Add some see-also references based on TFOT.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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5b168ee34b |
Merge changes from team/group/appdocsxml
This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
84282953d8 |
Document that the duration of dtmf may be passed to
the SendDTMF application. Also correct the default pause between digits. (closes issue #13102) Reported by: eliel Patches: app_senddtmf.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
51391e6b09 |
shuffle a little bit the content of header files to reduce dependencies.
In this commit: - move the ast_register/unregister_app functions to module.h to avoid the need to include pbx.h for the simpler apps; - move the ast_group structure to channel.h to remove the dependency of app.h on linkedlists.h Note, this is a long process that I am doing in small steps. The main difficulty is that now for each subsystem we have a single header (e.g. channel.h) included by the subsystem provider (usually one file, e.g. channel.c) and by its clients (dozens of them, e.g. we have some 70+ apps and 30+ functions). This requires the clients to include all the extra headers required by the provider (eg. lock.h, linkedlists.h, definitions of substructures...) even though many of the clients would be just happy with opaque struct declarations and function prototypes. The long term plan is to eventually rectify this structure so that the compilation can become faster, and also APIs are more stable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
ea2c54859d |
more removal of redundant headers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89519 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
7e8835e0d7 |
remove another set of redundant #include "asterisk/options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0595b5e2aa |
include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fdb7f7ba3d |
Start untangling header inclusion in a way that does not affect
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
09d9cce1d8 |
access channel locks through ast_channel_lock/unlock/trylock and not
through ast_mutex primitives. To detect all occurrences, I have renamed the lock field in struct ast_channel so it is clear that it shouldn't be used directly. There are some uses in res/res_features.c (see details of the diff) that are error prone as they try and lock two channels without caring about the order (or without explaining why it is safe). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89293 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
ac6c35c2a6 |
Fix bug where a NULL timeout would make things explode if SendDTMF was called with it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78279 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
9ef1b0a974 |
Extend the ast_senddigit and ast_dtmf_stream API calls to allow the duration of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
354b6dcf41 |
umm... let's build with --enable-dev-mode, mmkay?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77819 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
20bbd09de3 |
Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
b8cd949cce |
Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
96a646734f |
It is no longer required for each module that deals with a channel to call ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
70b945c4f1 |
Merged revisions 50073 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.4 ........ r50073 | mogorman | 2007-01-08 15:11:16 -0600 (Mon, 08 Jan 2007) | 1 line we can't unlock a channel if we cant find it. - AnthonyL bug #8741 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50074 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
87b9abc892 |
Merged revisions 49676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r49676 | kpfleming | 2007-01-05 16:16:33 -0600 (Fri, 05 Jan 2007) | 2 lines reduce stack consumption for AMI and AMI/HTTP requests by nearly 20K in most cases ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49678 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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0a27d8bfe5 |
merge new_loader_completion branch, including (at least):
- restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
472c1ca282 |
simplify autoconfig include mechanism (make tholo happy he can use lint again :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
04ecb29d03 |
remove almost all of the checks of the result from ast_strdupa() or alloca().
As it turns out, all of these checks were useless, because alloca will never return NULL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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f4f7b74655 |
- formatting fixes
- don't declare a variable in the middle of a block - keep track of the result of registration functions in load_module() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24805 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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7a9472f8a4 |
New manager action SendDTMF #6682 (squinky86)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
e43bc6634d |
This rather large commit changes the way modules are loaded.
As partly documented in loader.c and include/asterisk/module.h, modules are now expected to return all of their methods and flags into a structure 'mod_data', and are normally loaded with RTLD_NOW | RTLD_LOCAL, so symbols are resolved immediately and conflicts should be less likely. Only in a small number of cases (res_*, typically) modules are loaded RTLD_GLOBAL, so they can export symbols. The core of the change is only the two files loader.c and include/asterisk/module.h, all the rest is simply adaptation of the existing modules to the new API, a rather mechanical (but believe me, time and finger-consuming!) process whose detail you can figure out by svn diff'ing any single module. Expect some minor compilation issue after this change, please report it on mantis http://bugs.digium.com/view.php?id=6968 so we collect all the feedback in one place. I am just sorry that this change missed SVN version number 20000! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
f10f427d49 |
since the module API is changing, it's a good time to const-ify the description() and key() return values
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
28fd3a4ef1 |
Merged revisions 17283 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.2 ........ r17283 | mogorman | 2006-04-04 10:24:50 -0500 (Tue, 04 Apr 2006) | 3 lines Adds documentation to show what the w flag. Patch from Ian Kinner at Digium. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
a0d438fb6c |
remove the uses of the deprecated STANDARD_LOCAL_USER
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
|
4414f45393 |
on this pass, only remove duplicate log messages
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8403 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
|
9fa6eb5e07 |
revert my pass through the tree to remove checks of the result of ast_strdupa
(revisions 8378 through 8381) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8387 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
|
7ad681adc8 |
remove lots of useless checks of the result of ast_strdupa
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8379 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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a725468381 |
update doxygen docs to specify authors
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |