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Kevin P. Fleming 79087cbbd5 Ensure that all ast_datastore_info structures are 'const'. 
While addressing a bug, I came across a instance of 'struct ast_datastore_info'
that was not declared 'const'. Since the API already expects them to be
'const', this patch changes the declarations of all existing instances
that were not already declared that way.
........

Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
 		2012-07-18 17:18:20 +00:00
..
h323 Asterisk media architecture conversion - no more format bitfields 2011-02-03 16:22:10 +00:00
misdn Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00
sip Add support for SIP over WebSocket. 2012-07-16 12:35:04 +00:00
Makefile Extract sig_ss7 out of chan_dahdi. 2010-06-07 20:04:42 +00:00
chan_agent.c Add some additional documentation for core AMI events 2012-07-10 22:26:27 +00:00
chan_alsa.c Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 2012-05-14 19:44:27 +00:00
chan_bridge.c Finalize ast_channel opaquification 2012-03-13 18:20:34 +00:00
chan_console.c Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 2012-05-14 19:44:27 +00:00
chan_dahdi.c Add missing ast_hangup() calls on some analog exception paths. 2012-07-12 20:28:07 +00:00
chan_gtalk.c Add a new unified Jingle, Google Jingle, and Google Talk channel driver written from scratch called chan_motif. 2012-07-07 17:06:51 +00:00
chan_h323.c Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 2012-05-14 19:44:27 +00:00
chan_iax2.c Ensure that all ast_datastore_info structures are 'const'. 2012-07-18 17:18:20 +00:00
chan_jingle.c Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 2012-05-14 19:44:27 +00:00
chan_local.c Add some additional documentation for core AMI events 2012-07-10 22:26:27 +00:00
chan_mgcp.c Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 2012-05-14 19:44:27 +00:00
chan_misdn.c Fix chan_misdn compile error. 2012-06-20 21:32:40 +00:00
chan_motif.c Do not consider failure to read the configuration file in chan_motif to be a show stopper for loading Asterisk by returning decline instead of failure. 2012-07-11 16:42:01 +00:00
chan_multicast_rtp.c Opaquification for ast_format structs in struct ast_channel 2012-02-24 00:32:20 +00:00
chan_nbs.c Opaquification for ast_format structs in struct ast_channel 2012-02-24 00:32:20 +00:00
chan_oss.c Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 2012-05-14 19:44:27 +00:00
chan_phone.c Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 2012-05-14 19:44:27 +00:00
chan_phone.h rename DialTone.h to chan_phone.h because chan_phone.c is the only file using it 2008-05-28 20:00:21 +00:00
chan_sip.c Fix a crash occurring as a result of excess stack usage. 2012-07-18 11:38:05 +00:00
chan_sip.exports.in Merge changes dealing with support for Digium phones. 2012-06-04 20:26:12 +00:00
chan_skinny.c Various small chan_skinny fixes and cleanup 2012-06-15 21:18:56 +00:00
chan_unistim.c Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant. 2012-07-18 07:17:00 +00:00
chan_vpb.cc Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) 2012-05-14 19:44:27 +00:00
console_board.c Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00
console_gui.c Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00
console_video.c Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00
console_video.h implement the 'freeze' function for incoming frames; 2008-06-30 15:45:15 +00:00
iax2-parser.c Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00
iax2-parser.h Asterisk media architecture conversion - no more format bitfields 2011-02-03 16:22:10 +00:00
iax2-provision.c Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00
iax2-provision.h Don't reload a configuration file if nothing has changed. 2007-08-16 21:09:46 +00:00
iax2.h Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff 2011-02-22 23:04:49 +00:00
misdn_config.c Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00
sig_analog.c Add missing ast_hangup() calls on some analog exception paths. 2012-07-12 20:28:07 +00:00
sig_analog.h push 'outgoing' flag from sig_XXX up to chan_dahdi 2012-02-18 08:02:08 +00:00
sig_pri.c Fix validation errors when producing documentation using default build script 2012-07-11 02:06:05 +00:00
sig_pri.h Clear ISDN channel resetting state if the peer continues to use it. 2012-04-25 19:55:12 +00:00
sig_ss7.c Remove unnecessary generation of informational cause frames 2012-07-06 22:03:44 +00:00
sig_ss7.h Setup DSP when SS7 call is connected or early media is available. 2012-03-05 21:48:32 +00:00
vcodecs.c Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00
vgrabbers.c Multiple revisions 369001-369002 2012-06-15 16:20:16 +00:00