111 Commits (master)

Author	SHA1	Message	Date
Sean Bright	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
George Joseph	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
George Joseph	c2ffecfc66	Fix application references to Background ... The app is actually named "BackGround" but several references in XML documentation were spelled "Background" with the lower case "g". This was causing documentation links to return "not found" messages.	7 months ago
Sean Bright	ca931c9436	app.c: Allow ampersands in playback lists to be escaped. ... Any function or application that accepts a `&`-separated list of filenames can now include a literal `&` in a filename by wrapping the entire filename in single quotes, e.g.: ``` exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world) ``` Fixes #172 UpgradeNote: Ampersands in URLs passed to the `Playback()`, `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()` applications as filename arguments can now be escaped by single quoting the filename. Additionally, this is also possible when using the `CONFBRIDGE` dialplan function, or configuring various features in `confbridge.conf` and `queues.conf`.	1 year ago
Sean Bright	cbaba132a7	app_playback.c: Fix PLAYBACKSTATUS regression. ... In Asterisk 11, if a channel was redirected away during Playback(), the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12 (specifically commit 7d9871b394) that behavior was inadvertently changed and the same operation would result in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11 behavior has been restored. Partial fix for ASTERISK~25661. Change-Id: I53f54e56b59b61c99403a481b6cb8d88b5a559ff	2 years ago
Naveen Albert	bcc18ca9f5	general: Fix various typos. ... ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Josh Soref	eb03b18ff9	apps: Spelling fixes ... Correct typos of the following word families: simultaneously administrator directforward attachfmt dailplan automatically applicable nouns explicit outside sponsored attachment audio spied doesn't counting encoded implements recursively emailaddress arguments queuerules members priority output advanced silencethreshold brazilian debugging argument meadmin formatting integrated sneakiness ASTERISK-29714 Change-Id: Ie5ecaec91c00b26309da4e51cfc0991a5bb7d092	3 years ago
Shloime Rosenblum	cfae5224e3	apps/app_playback.c: Add 'mix' option to app_playback ... I am adding a mix option that will play by filename and say.conf unlike say option that will only play with say.conf. It will look on the format of the name, if it is like say it play with say.conf if not it will play the file name. ASTERISK-29662 Change-Id: I815816916a308f0fa8f165140dc15772dcbd547a	4 years ago
Kevin Harwell	bdd785d31c	various files - fix some alerts raised by lgtm code analysis ... This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	6 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	7 years ago
Holger Hans Peter Freyther	b54eb167b4	app_playback.c: Use the timezonename parameter ... In say_date_generic the timezonename parameter is passed but never used. Fix it by passing it to the ast_localtime function. ASTERISK-27124 Change-Id: I63106b8db10426d417d7275f22554a616e92fae4	8 years ago
Corey Farrell	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
George Joseph	31f0d78d7b	app_playback: Suppress warnings on playback if channel hung up ... If a channel hangs up while an audio file is playing, there's no need to clutter up the logs with a warning so suppress it if ast_check_hangup returns true. Also, change warning to debug/2 in file.c if writing a frame fails. Same reasoning. Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89 Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Rodrigo Ramírez Norambuena	eec010829a	AST_MODULE_INFO: Format corrections to the usages of AST_MODULE_INFO macro. ... Change-Id: Icf88f9f861c6b2a16e5f626ff25795218a6f2723	10 years ago
Matt Jordan	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
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ... ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	ccdfe67bf2	Check result of ast_var_assign() calls for memory allocation failure. ... We try to keep the system running even when all available memory is spent. Review: https://reviewboard.asterisk.org/r/2734/ ........ Merged revisions 396279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396287 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	b607a2dbf9	Properly load say.conf upon reload of module app_playback. ... If say.conf did not exists prior to originally loading module app_playback it would not load on subsequent reloads of the module once it had been created. This occurred because upon reload of the app_playback module it would only load a new configuration if an old one had previously existed. This fix simply removed the association between checking if an old configuration existed and the loading of the new one. (closes issue ASTERISK-20800) Reported by: pgoergler ........ Merged revisions 381216 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 381217 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7d9871b394	Add ControlPlayback manager action ... This patch adds the capability for asynchronous manipulation of audio being played back to a channel though a new AMI action "ControlPlayback". The ControlPlayback action supports a number of operations, the availability of which depend on the application being used to send audio to the channel. When the audio playback was initiated using the ControlPlayback application or CONTROL STREAM FILE AGI command, the audio can be paused, stopped, restarted, reversed, or skipped forward. When initiated by other mechanisms (such as the Playback application), the audio can be stopped, reversed, or skipped forward. Review: https://reviewboard.asterisk.org/r/2265/ (closes issue ASTERISK-20882) Reported by: mjordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types ... Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields ... Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions ... There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Leif Madsen	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
Olle Johansson	eecf1978af	Add doxygen documentation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1cf5422dc8	Merged revisions 220288 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines Implicitly sending a progress signal breaks some applications. Call Progress() in your dialplan if you explicitly want progress to be sent. (Reverts change 216430, closes issue #15957) Reported by: Pavel Troller on the Asterisk-Dev mailing list http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@220289 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Olle Johansson	98f18d56b8	Merged revisions 216430 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines Make apps send PROGRESS control frame for early media and fix too early media issue in SIP The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI links *before* any call progress. The SIP channel receives these frames and by default signals 183 Session progress and starts sending media. This will cause phones to play silence and ignore the later 180 ringing message. A bad user experience. The fix is twofold: - We discovered that asterisk apps that support early media ("noanswer") did not send any PROGRESS frame to indicate early media. Fixed. - We introduce a setting in chan_sip so that users can disable any relay of media frames before the outbound channel actually indicates any sort of call progress. In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions of Asterisk, this will be enabled. We don't assume that it will change your Asterisk phone experience - only for the better. We encourage third-party application developers to make sure that if they have applications that wants to send early media, add a PROGRESS control frame transmission to make sure that all channel drivers actually will start sending early media. This has not been the default in Asterisk previous to this patch, so if you got inspiration from our code, you need to update accordingly. Sorry for the trouble and thanks for your support. This code has been running for a few months in a large scale installation (over 250 servers with PRI and/or BRI links to old PBX systems). That's no proof that this is an excellent patch, but, well, it's tested :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) ... This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	1c988d8996	add 'const' qualifiers in various places where they should have been ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	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
Russell Bryant	6f314f4d42	Fix various spelling and grammatical issues in documentation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	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
Steve Murphy	e74584ca3c	(closes issue #13557) ... Reported by: nickpeirson The user attached a patch, but the license is not yet recorded. I took the liberty of finding and replacing ALL index() calls with strchr() calls, and that involves more than just main/pbx.c; chan_oss, app_playback, func_cut also had calls to index(), and I changed them out. 1.4 had no references to index() at all. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	08af5bb312	Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating ... when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3ffb39833b	More RSW merges. Everything from apps/ except for the big offenders ... app_voicemail and app_queue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137055 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	463a5dbd0a	Whitespace changes only ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	6928ccfa02	Merged revisions 111391 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines These small documentation updates made in response to a query in asterisk-users, where a user was using Playback, but needed the features of Background, and had no idea that Background existed, or that it might provide the features he needed. I thought the best way to avert these kinds of queries was to provide "See Also" references in all three of "Background", "Playback", "WaitExten". Perhaps a project to do this with all related apps is in order. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1ec8cb41a8	Merge changes from team/mvanbaak/cli-command-audit ... (closes issue #8925) About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI commands in Asterisk 1.4 for the next version of their book, they documented a lot of inconsistencies. This set of changes addresses all of these issues and has been reviewed by Leif. While this does introduce even more changes to the CLI command structure, it makes everything consistent, which is the most important thing. Thanks to all that helped with this one! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	5f3a28e588	Merged revisions 89618 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines After issuing a "say load new", if a caller hangs up during the middle of playback of a number, app_playback will continue to try to play the remaining files. With this change, no more files will be played back upon hangup. (closes issue #11345, reported and patched by IgorG) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	b1fe2d85d3	remove redundant headers ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	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
Luigi Rizzo	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
Joshua Colp	23cfef1cc9	Change warning messages (which are really debug messages) into debug messages. ... (closes issue #11288) Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	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
Luigi Rizzo	7f8ecd2cd3	make the 'name' and 'value' fields in ast_variable const char * ... This prevents modifying the strings in the stored variables, and catched a few instances where this was actually done. Given the differences between trunk and 1.4 (and the fact that this is effectively an API change) it is better to fix 1.4 independently. These are chan_sip.c::sip_register() chan_skinny.c:: near line 2847 config.c:: near line 1774 logger.c::make_components() res_adsi.c:: near line 1049 I may have missed some instances for modules that do not build here. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	6f3f010ca0	Suppress erroneous warnings on load. ... Reported by: eliel Patch by: eliel Closes issue #11177 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89081 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Mark Michelson	5a4867543d	"show application <foo>" changes for clarity. ... (closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Steve Murphy	63f2f04cf4	This commits the performance mods that give the priority processing engine in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	39ce390203	Convert some spaces to tabs and make it so the CLI command is only registered ... once instead of 3 times. (closes issue #11053) Reported by: seanbright Patches: app_playback.patch uploaded by seanbright (license 71) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	ebe4050128	Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	b0f3e6097e	Convert NEW_CLI to AST_CLI. ... Closes issue #11039, as suggested by seanbright. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago