aboutsummaryrefslogtreecommitdiffstats
path: root/apps/app_playback.c
AgeCommit message (Collapse)AuthorFilesLines
2009-09-24Merged revisions 220289 via svnmerge from tilghman1-3/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.2@220292 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-07Merged revisions 216438 via svnmerge from oej1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.2@216647 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Janitor, use ARRAY_LEN() when possible.eliel1-2/+2
(closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02Fix various spelling and grammatical issues in documentationrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153468 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01Merge changes from team/group/appdocsxmlrussell1-20/+43
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: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-25(closes issue #13557)murf1-4/+4
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: http://svn.digium.com/svn/asterisk/trunk@144569 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman1-2/+6
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: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-10More RSW merges. Everything from apps/ except for the big offendersseanbright1-5/+5
app_voicemail and app_queue. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137055 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-25Whitespace changes onlytilghman1-46/+46
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114667 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-27Merged revisions 111391 via svnmerge from murf1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@111410 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-08Merge changes from team/mvanbaak/cli-command-auditrussell1-1/+1
(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: http://svn.digium.com/svn/asterisk/trunk@103171 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Merged revisions 89618 via svnmerge from mmichelson1-1/+5
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: http://svn.digium.com/svn/asterisk/trunk@89619 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22remove redundant headersrizzo1-6/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89518 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-1/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19Change warning messages (which are really debug messages) into debug messages.file1-4/+4
(closes issue #11288) Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded by IgorG (license 20) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89410 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-4/+0
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: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-14make the 'name' and 'value' fields in ast_variable const char *rizzo1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@89268 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-07Suppress erroneous warnings on load.tilghman1-4/+4
Reported by: eliel Patch by: eliel Closes issue #11177 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89081 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06"show application <foo>" changes for clarity.mmichelson1-2/+2
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89044 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01This commits the performance mods that give the priority processing engine ↵murf1-1/+0
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: http://svn.digium.com/svn/asterisk/trunk@88166 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22Convert some spaces to tabs and make it so the CLI command is only registeredrussell1-15/+10
once instead of 3 times. (closes issue #11053) Reported by: seanbright Patches: app_playback.patch uploaded by seanbright (license 71) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86835 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵qwell1-3/+3
didn't make much sense git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Convert NEW_CLI to AST_CLI.qwell1-3/+3
Closes issue #11039, as suggested by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-20More NEW_CLI conversions.qwell1-30/+26
(issue #10724) Patches: app_playback.c.patch uploaded by moy (license 222) app_minivm.c.patch uploaded by eliel (license 64) astmm.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@83381 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-16Don't reload a configuration file if nothing has changed.tilghman1-2/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79747 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵tilghman1-1/+1
few other formatting cleanups, too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-20Add support for default "say mode" (whether to use the "old" method or "new" ↵qwell1-22/+75
method. "new" method being config file) Add support for autocomplete of "say load" CLI command. Patch by IgorG (closes issue #10243) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76216 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-18Merge in ast_strftime branch, which changes timestamps to be accurate to the ↵tilghman1-2/+3
microsecond, instead of only to the second git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75706 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-4/+0
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16It is no longer required for each module that deals with a channel to call ↵file1-2/+0
ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-12Completely remove all of the code related to jumping to priority n + 101. yay!russell1-10/+2
(issue #9926, caio1982) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68970 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-07Merged revisions 53399 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53399 | file | 2007-02-07 12:04:44 -0500 (Wed, 07 Feb 2007) | 2 lines Directly load say.conf in load_module instead of calling the reload function. (issue #8946 reported by junky) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53400 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-05Merged revisions 53152 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53152 | file | 2007-02-05 11:06:18 -0600 (Mon, 05 Feb 2007) | 2 lines Ensure say_cfg is NULL when the module is loaded. (issue #8946 reported by junky) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53153 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-05Merged revisions 53150 via svnmerge from file1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53150 | file | 2007-02-05 10:02:00 -0600 (Mon, 05 Feb 2007) | 2 lines Unregister Playback CLI commands as well as dialplan application. (issue #8946 reported by junky) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53151 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-13Merged revisions 45051 via svnmerge from file1-5/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r45051 | file | 2006-10-13 12:20:58 -0400 (Fri, 13 Oct 2006) | 2 lines Move say.conf existence check to do_say function since it is called from multiple places (issue #8144 reported by kshumard) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45052 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-28Merged revisions 43933 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2 lines Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43934 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-27Merged revisions 43803 via svnmerge from qwell1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4 lines Fix an issue with PLAYBACKSTATUS not being set under certain circumstances. Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string. Fix Background() to return -1 like Playback(), if no args are specified. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43804 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18merge qwell's CLI verbification workkpfleming1-3/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43212 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21merge new_loader_completion branch, including (at least):kpfleming1-22/+13
- 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: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-13destroy the loaded say.conf on module unloadrussell1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33786 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-5/+4
again :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-29remove an explicit constant;rizzo1-1/+5
add a comment on the need to sort patterns in the standard way. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@30700 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10support reload say.conf to ease testingrizzo1-1/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26529 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10remove almost all of the checks of the result from ast_strdupa() or alloca().russell1-4/+1
As it turns out, all of these checks were useless, because alloca will never return NULL. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26451 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-19add experimental code for new-style "say" application.rizzo1-3/+323
The rules for spelling out numbers and dates are in the config file "say.conf", which can be edited to implement national or even local language rules. The new code can be enabled through the cli command 'say load new' while the old code can be restored with 'say load old' Eventually, this code should go to a better place, but for the time being we keep here as it provides very similar functions. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@21421 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-14This rather large commit changes the way modules are loaded. rizzo1-11/+6
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: http://svn.digium.com/svn/asterisk/trunk@20003 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-08since the module API is changing, it's a good time to const-ify the ↵kpfleming1-2/+2
description() and key() return values git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18552 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-15remove the uses of the deprecated STANDARD_LOCAL_USERrussell1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10241 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-21on this pass, only remove duplicate log messagesrussell1-4/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8403 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-24 01:28:29 +0000