asterisk.git - Unnamed repository; edit this file 'description' to name the repository.

Age	Commit message (Collapse)	Author
2009-04-29	Merge str_substitution branch.	Tilghman Lesher
	This branch adds additional methods to dialplan functions, whereby the result buffers are now dynamic buffers, which can be expanded to the size of any result. No longer are variable substitutions limited to 4095 bytes of data. In addition, the common case of needing buffers much smaller than that will enable substitution to only take up the amount of memory actually needed. The existing variable substitution routines are still available, but users of those API calls should transition to using the dynamic-buffer APIs. Reviewboard: http://reviewboard.digium.com/r/174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02	Fix various spelling and grammatical issues in documentation	Russell Bryant
	git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01	Merge changes from team/group/appdocsxml	Russell Bryant
	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
2008-08-05	make datastore creation and destruction a generic API since it is not really ↵	Kevin P. Fleming
	channel related, and add the ability to add/find/remove datastores to manager sessions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22	Fix a few places where frame data was used directly.	Jason Parker
	git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-13	Re-introduce proper error handling that was removed in recent commits.	Russell Bryant
	git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-10	since we unregister, that has not been properly registered, i standardized this.	Claude Patry
	git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-05	Merged revisions 115327 via svnmerge from	Joshua Colp
	https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115327 \| file \| 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) \| 2 lines Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-01	Add "read" capability to new libspeex functions in func_speex.c.	Brett Bryant
	func_speex.c is based on contributions from Switchvox. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-01	Add two new dialplan functions from libspeex for applying audio gain control	Brett Bryant
	and denoising to a channel, AGC() and DENOISE(). Also included, is a change to the audiohook API to add a new function (ast_audiohook_remove) that can remove an audiohook from a channel before it is detached. This code is based on a contribution from Switchvox. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114926 65c4cc65-6c06-0410-ace0-fbb531ad65f3