dect/asterisk - Asterisk with DECT support

Age	Commit message (Collapse)	Author	Files	Lines
2009-10-20	Merged revisions 224856 via svnmerge from	tilghman	1	-8/+13
	https://origsvn.digium.com/svn/asterisk/trunk ................ r224856 \| tilghman \| 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) \| 12 lines Merged revisions 224855 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 \| tilghman \| 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) \| 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@224859 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10	AST-2009-005	tilghman	1	-1/+1
	git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@211580 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-02	Fix various spelling and grammatical issues in documentation	russell	1	-3/+3
	git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153468 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01	Merge changes from team/group/appdocsxml	russell	1	-27/+46
	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
2008-08-05	make datastore creation and destruction a generic API since it is not really ↵	kpfleming	1	-4/+4
	channel related, and add the ability to add/find/remove datastores to manager sessions git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135680 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-22	Fix a few places where frame data was used directly.	qwell	1	-1/+1
	git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117828 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-13	Re-introduce proper error handling that was removed in recent commits.	russell	1	-4/+8
	git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115850 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-10	since we unregister, that has not been properly registered, i standardized this.	junky	1	-8/+4
	git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115593 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-05	Merged revisions 115327 via svnmerge from	file	1	-0/+2
	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: http://svn.digium.com/svn/asterisk/trunk@115328 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01	Add "read" capability to new libspeex functions in func_speex.c.	bbryant	1	-1/+40
	func_speex.c is based on contributions from Switchvox. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114977 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01	Add two new dialplan functions from libspeex for applying audio gain control	bbryant	1	-0/+310
	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: http://svn.digium.com/svn/asterisk/trunk@114926 f38db490-d61c-443f-a65b-d21fe96a405b