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${ noResults }
72 Commits (37ece75677bff5655f0a3fb4dbda2b7c4847bc50)
Author | SHA1 | Message | Date |
---|---|---|---|
|
1633410161 |
res_stir_shaken: refactor utility function
Refactors temp file utility function into file.c. ASTERISK-29809 #close Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b |
4 years ago |
|
9440f6ec58 |
main: Fix for Doxygen.
ASTERISK-29763 Change-Id: Ib8359e3590a9109eb04a5376559d040e5e21867e |
4 years ago |
|
d374d63ef8 |
app_voicemail: Refactor email generation functions
Refactors generic functions used for email generation into utils.c so that they can be used by multiple modules, including app_voicemail and app_minivm, to avoid code duplication. ASTERISK-29715 #close Change-Id: I1de0ed3483623e9599711129edc817c45ad237ee |
4 years ago |
|
5d3a115bee |
include: Spelling fixes
Correct typos of the following word families: activities forward occurs unprepared association compress extracted doubly callback prometheus underlying keyframe continue convenience calculates ignorepattern determine subscribers subsystem synthetic applies example manager established result microseconds occurrences unsuccessful accommodates related signifying unsubscribe greater fastforward itself unregistering using translator sorcery implementation serializers asynchronous unknowingly initialization determining category these persistent propagate outputted string allocated decremented second cacheability destructor impaired decrypted relies signaling based suspended retrieved functions search auth considered ASTERISK-29714 Change-Id: I542ce887a16603f886a915920d5710d4a0a1358d |
4 years ago |
|
0827d5cc53 |
Add the ability to read the media file type from HTTP header for playback
How it works today: media_cache tries to parse out the extension of the media file to be played from the URI provided to Asterisk while caching the file. What's expected: Better will be to have Asterisk get extension from other ways too. One of the common ways is to get the type of content from the CONTENT-TYPE header in the HTTP response for fetching the media file using the URI provided. Steps to Reproduce: Provide a URL of the form: http://host/media/1234 to Asterisk for media playback. It fails to play and logs show the following error line: [Sep 15 15:48:05] WARNING [29148] [C-00000092] file.c: File http://host/media/1234 does not exist in any format Scenario this issue is blocking: In the case where the media files are stored in some cloud object store, following can block the media being played via Asterisk: Cloud storage generally needs authenticated access to the storage. The way to do that is by using signed URIs. With the signed URIs there's no way to preserve the name of the file. In most cases Cloud storage returns a key to access the object and preserving file name is also not a thing there ASTERISK-27286 Reporter: Gaurav Khurana Change-Id: I1b14692a49b2c1ac67688f58757184122e92ba89 |
8 years ago |
|
84a6365164 |
loader: Use ast_cli_completion_add for 'module load' completion.
This addresses all performance issues with 'module load' completion. In addition to using ast_cli_completion_add we stop using libedit's filename_completion_function, instead using ast_file_read_dir. This ensures all results are produced from a single call to opendir. Change-Id: I8bf51ffaa7ef1606f3bd1b5bb13f1905d72c6134 |
8 years ago |
|
fd0ca1c3f9 |
Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0 |
8 years ago |
|
70d5f90e3d |
stasis_recording/stored: remove calls to deprecated readdir_r function.
The readdir_r function has been deprecated and should no longer be used. This patch removes the readdir_r dependency (replaced it with readdir) and also moves the directory search code to a more centralized spot (file.c) Also removed a strict dependency on the dirent structure's d_type field as it is not portable. The code now checks to see if the value is available. If so, it tries to use it, but defaults back to using the stats function if necessary. Lastly, for most implementations of readdir it *should* be thread-safe to make concurrent calls to it as long as different directory streams are specified. glibc falls into this category. However, since it is possible that there exist some implementations that are not safe, locking has been added for those other than glibc. ASTERISK-26412 ASTERISK-26509 #close Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba |
9 years ago |
|
662b687dbe |
Voicemail: get correct duration when copying file to vm
Changes made during format improvements resulted in the recording to voicemail option 'm' of the MixMonitor app writing a zero length duration in the msgXXXX.txt file. This change introduces a new function ast_ratestream(), which provides the sample rate of the format associated with the stream, and updates the app_voicemail function for ast_app_copy_recording_to_vm to calculate the right duration. Review: https://reviewboard.asterisk.org/r/3996/ ASTERISK-24328 #close ........ Merged revisions 423192 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
a2c912e997 |
media formats: re-architect handling of media for performance improvements
In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
a75fd32212 |
ARI - channel recording support
This patch is the first step in adding recording support to the Asterisk REST Interface. Recordings are stored in /var/spool/recording. Since recordings may be destructive (overwriting existing files), the API rejects attempts to escape the recording directory (avoiding issues if someone attempts to record to ../../lib/sounds/greeting, for example). (closes issue ASTERISK-21594) (closes issue ASTERISK-21581) Review: https://reviewboard.asterisk.org/r/2612/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
a0b7a49a4a |
Index installed sounds and implement ARI sounds queries
This adds support for stasis/sounds and stasis/sounds/{ID} queries via the Asterisk RESTful Interface (ARI, formerly Stasis-HTTP). The following changes have been made to accomplish this: * A modular indexer was created for local media. * A new function to get an ast_format associated with a file extension was added. * Modifications were made to the built-in HTTP server so that URI decoding could be deferred to the URI handler when necessary. * The Stasis-HTTP sounds JSON documentation was modified to handle cases where multiple languages are installed in different formats. * Register and Unregister events for formats were added to the system topic. (closes issue ASTERISK-21584) (closes issue ASTERISK-21585) Review: https://reviewboard.asterisk.org/r/2507/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
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 |
13 years ago |
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14a985560e |
Merge changes dealing with support for Digium phones.
Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
518ccb6706 |
Remove autojump extensions from SayUnixTime, make an option to perform automatic jumps.
When a caller sends DTMF while the SayUnixTime application is saying the time, The call would jump to the next extension much like it does during Background(). This patch adds option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch allows arguments to sayunixtime to not be used as empty strings in the case of something like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue ASTERISK-16675) Reported by: jlpedrosa Patches: patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959) Review: https://reviewboard.asterisk.org/r/956/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
c26c190711 |
Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
1225ee831c |
Merged revisions 265089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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31848bcdd1 |
Merged revisions 231614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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936a2bd202 |
Reverted 231616
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8d1f4fa5ea |
Merged revisions 231614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231616 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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dd50b9e8b5 |
Merged revisions 222878 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222880 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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3a9c27459e |
Merged revisions 158072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r158072 | twilson | 2008-11-20 11:48:58 -0600 (Thu, 20 Nov 2008) | 2 lines Begin on a crusade to end trailing whitespace! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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28705e1bba |
Typo fixed earlier, that wasn't a typo after all.
Didn't a clever guy once say "Compile before you commit" ? :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93152 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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799fd65435 |
fix a typo from revision 93138
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93151 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2ee2c0b3a4 |
include mmap header if detected by configure
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93138 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8a3e984b0c |
Merged revisions 90142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | 4 lines Merge a change from team/russell/chan_refcount ... This makes ast_stopstream() thread-safe. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90143 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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730e4eaca4 |
implement the split of file.h and mod_format.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89515 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 |
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a6507769ca |
Merged revisions 85316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10 Oct 2007) | 6 lines I introduced a new member to the ast_filestream struct in 1.4.12, but put it in the middle of the struct, instead of at the end. One of the Debian folks, paravoid, pointed out that this breaks binary compatability with modules compiled against older headers. So, I'm moving the new member to the end of the struct to resolve the situation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6e8997fe3c |
Change function documentation to use doxygen tags. (Really, I just needed
to make some minor change in trunk to test something with automerge ...) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83849 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
8bcfddc8ec |
Merged revisions 81599 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | 11 lines Fix an issue that can occur when you do an attended transfer to parking. If you complete the transfer before the announcement of the parking spot finishes, then the channel being parked will hear the remainder of the announcement. These changes make it so that will not happen anymore. Basically, res_features sets a flag on the channel is playing the announcement to so that the file streaming core knows that it needs to watch out for a channel masquerade, and if it occurs, to abort the announcement. (closes BE-182) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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eaebbdab61 |
(closes issue #10271)
Reported by: snuffy Patches: doxygen-updates.diff uploaded by snuffy (license 35) Another big batch of doxygen documentation updates git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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8d1e53958c |
Merge a bunch of doxygen updates to header files. This includes changes to
use the \retval tag for documenting return values, fixing various warnings when generating the documentation, and various other things. (closes issue #10203, snuffy) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75164 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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6dcb17baaf |
This update fulfils the request of bug 7109, which claimed the language arg to ast_stream_and_wait() was redundant. Almost all calls just used chan->language, and seeing how chan is the first argument, this certainly seems redundant. A change of language could just as easily be done by simply changing the channel language before calling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47821 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
19 years ago |
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8a2e564df5 |
Issue 8246 Doxygen updates (kshumard)
THANK YOU! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46434 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 |
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427df3f6c3 |
yet another massive performance and memory savings improvement
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32349 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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7e0d402e88 |
move wait_and_stream to ast_wait_and_stream() because equivalent
code is replicated in way too many places not to have a global function for that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22075 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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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 |
20 years ago |
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4beb6deeaa |
Largely simplify format handlers (for file copy etc.)
collecting common functions in a single place and removing them from the individual handlers. The full description is on mantis, http://bugs.digium.com/view.php?id=6375 and only the ogg_vorbis handler needs to be converted to the new structure. As a result of this change, format_au.c and format_pcm_alaw.c should go away (in a separate commit) as their functionality (trivial) has been merged in another file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17243 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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aa20c556f7 |
Bug 5984 - Convert file offsets to 64 bit
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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2c65582b66 |
remove extraneous svn:executable properties
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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0aa9429c96 |
issue #5658, different fix
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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3332a8acd1 |
clean up a lot of doxygen errors and warnings (issue #5522)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6865 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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3453e3efa5 |
Doxygen documentation update from oej (issue #5505)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6847 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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d499a85f05 |
Use FILE * instead of fd for files to support buffering
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6801 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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1632d52795 |
major header file cleanup: license, copyrights, descriptions, markers, etc.
remove deprecated config_old.c/config_old.h remove unused cvsid.h git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6454 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
20 years ago |
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97671a2b05 |
add 'exit context' and 'only stop on match' options to Background app (bug #4511)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
21 years ago |
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7a2d5cbb3e |
make ast_waitstream_* return value compatible with platforms that use unsigned char by default (bug #4455)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5846 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
21 years ago |
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1f9ab2380a |
use double-quotes instead of angle-brackets for non-system include files (bug #4058)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
21 years ago |