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2011-02-18Merged revisions 308242 via svnmerge from may1-5/+7
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 lines added g729onlyA option for announce only AnnexA g.729 codec in h.323 capabilities. Option can be global or per user/peer. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308243 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-12change malloc to ast_calloc calls to prevent crash of asteriskmay1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307677 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Asterisk media architecture conversion - no more format bitfieldsdvossel1-52/+53
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: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-25Use "local" instead of "system" header file inclusion.kpfleming1-4/+4
Now that these files are in the tree, they should prefer the tree's local copy of all Asterisk headers over any that may be installed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254931 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-01More 32->64 bit codec conversions.tilghman1-3/+3
In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231850 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Reworked chan_ooh323 channel module.may1-25/+375
Many architectural and functional changes. Main changes are threading model chanes (many thread in ooh323 stack instead of one), modifications and improvements in signalling part, additional codecs support (726, speex), t38 mode support. This module tested and used in production environment. (closes issue #15285) Reported by: may213 Tested by: sles, c0w, OrNix Review: https://reviewboard.asterisk.org/r/324/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227898 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-30Move Asterisk-addons modules into the main Asterisk source tree.russell1-0/+361
Someone asked yesterday, "is there a good reason why we can't just put these modules in Asterisk?". After a brief discussion, as long as the modules are clearly set aside in their own directory and not enabled by default, it is perfectly fine. For more information about why a module goes in addons, see README-addons.txt. chan_ooh323 does not currently compile as it is behind some trunk API updates. However, it will not build by default, so it should be okay for now. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204413 f38db490-d61c-443f-a65b-d21fe96a405b