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2009-02-19Merged revisions 177384 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r177384 | file | 2009-02-19 12:38:41 -0400 (Thu, 19 Feb 2009) | 10 lines Merged revisions 177383 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177383 | file | 2009-02-19 12:37:25 -0400 (Thu, 19 Feb 2009) | 3 lines If we are able to create a speech structure unset the ERROR variable in case it was previously set. (issue #LUMENVOX-13) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@177385 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-02Merged revisions ↵tilghman1-0/+1
147518,147689,148000,148112,148268,148917,148988,149062,149131,149201,149205,149208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r147518 | file | 2008-10-08 09:53:51 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions 147517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147517 | file | 2008-10-08 11:51:42 -0300 (Wed, 08 Oct 2008) | 2 lines If we receive DTMF make sure that the state of the speech structure goes back to being not ready. (issue #LUMENVOX-8) ........ ................ r147689 | kpfleming | 2008-10-08 17:26:55 -0500 (Wed, 08 Oct 2008) | 9 lines Merged revisions 147681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147681 | kpfleming | 2008-10-08 17:22:09 -0500 (Wed, 08 Oct 2008) | 3 lines when parsing a text configuration option, ensure that the buffer on the stack is actually large enough to hold the legal values of that option, and also ensure that sscanf() knows to stop parsing if it would overrun the buffer (without these changes, specifying "buffers=...,immediate" would overflow the buffer on the stack, and could not have worked as expected) ........ ................ r148000 | tilghman | 2008-10-09 14:39:34 -0500 (Thu, 09 Oct 2008) | 11 lines Merged revisions 147997 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r147997 | tilghman | 2008-10-09 14:38:33 -0500 (Thu, 09 Oct 2008) | 4 lines When blank, callerid name and number should display "unknown caller" in voicemail emails. (Closes issue #13643) ........ ................ r148112 | mmichelson | 2008-10-09 18:15:33 -0500 (Thu, 09 Oct 2008) | 26 lines Merged revisions 146026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r146026 | murf | 2008-10-03 12:12:54 -0500 (Fri, 03 Oct 2008) | 18 lines (closes issue #13579) Reported by: dwagner (closes issue #13584) Reported by: dwagner Tested by: murf, putnopvut The thought occurred to me that the res= from the extension spawn was ending up being returned from the bridge. "Thou shalt not poison the return value". Made the change and it appears to allow blind xfers to work as normal. If I'm wrong, reopen the bugs. But it looks good to me! Many thanks to putnopvut for helping me reproduce this! ........ ................ r148268 | tilghman | 2008-10-10 11:31:31 -0500 (Fri, 10 Oct 2008) | 14 lines Merged revisions 148257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148257 | tilghman | 2008-10-10 11:25:31 -0500 (Fri, 10 Oct 2008) | 7 lines User not notified of temporary greeting, if ODBC storage is in use. (closes issue #13659) Reported by: moliveras Patches: 20081009__bug13659.diff.txt uploaded by Corydon76 (license 14) Tested by: moliveras ........ ................ r148917 | tilghman | 2008-10-14 12:46:48 -0500 (Tue, 14 Oct 2008) | 11 lines Merged revisions 148916 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148916 | tilghman | 2008-10-14 12:41:08 -0500 (Tue, 14 Oct 2008) | 4 lines Ensure that mail headers are 7-bit clean, even when UTF-8 characters are used in headers like 'Subject' and 'To'. Closes AST-107. ........ ................ r148988 | tilghman | 2008-10-14 14:03:44 -0500 (Tue, 14 Oct 2008) | 9 lines Merged revisions 148987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r148987 | tilghman | 2008-10-14 14:03:08 -0500 (Tue, 14 Oct 2008) | 2 lines Some compilers warn, some don't. Fixing. ........ ................ r149062 | tilghman | 2008-10-14 15:16:48 -0500 (Tue, 14 Oct 2008) | 13 lines Merged revisions 149061 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149061 | tilghman | 2008-10-14 15:09:06 -0500 (Tue, 14 Oct 2008) | 6 lines Check correct values in the return of ast_waitfor(); also, get rid of a possible memory leak. (closes issue #13658) Reported by: explidous Patch by: me ........ ................ r149131 | mmichelson | 2008-10-14 16:08:48 -0500 (Tue, 14 Oct 2008) | 15 lines Merged revisions 149130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149130 | mmichelson | 2008-10-14 15:49:02 -0500 (Tue, 14 Oct 2008) | 7 lines Don't allow reserved characters to be used in register lines in sip.conf. (closes issue #13570) Reported by: putnopvut ........ ................ r149201 | mmichelson | 2008-10-14 17:41:13 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149200 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines Update the queue with the correct number of calls and whether the call was completed within the service level when a transfer takes place. This way, we do not "break" the leastrecent and fewestcalls strategies by not logging a call until after the transferred call has ended. (closes issue #13395) Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded by Marquis (license 32) ........ ................ r149205 | mmichelson | 2008-10-14 18:04:44 -0500 (Tue, 14 Oct 2008) | 20 lines Merged revisions 149204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149204 | mmichelson | 2008-10-14 18:00:01 -0500 (Tue, 14 Oct 2008) | 12 lines Add a tolerance period for sync-triggered audiohooks so that if packetization of audio is close (but not equal) we don't end up flushing the audiohooks over small inconsistencies in synchronization. Related to issue #13005, and solves the issue for most people who were experiencing the problem. However, a small number of people are still experiencing the problem on long calls, so I am not closing the issue yet ........ ................ r149208 | mmichelson | 2008-10-14 18:15:04 -0500 (Tue, 14 Oct 2008) | 17 lines Merged revisions 149207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r149207 | mmichelson | 2008-10-14 18:10:26 -0500 (Tue, 14 Oct 2008) | 9 lines Call register_peer_exten even in the case that the peer's IP/port does not change. (closes issue #13309) Reported by: dimas Patches: v2-13309.patch uploaded by dimas (license 88) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@160387 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-07Merged revisions 106553 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r106553 | tilghman | 2008-03-07 00:54:47 -0600 (Fri, 07 Mar 2008) | 14 lines Merged revisions 106552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines Safely use the strncat() function. (closes issue #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt uploaded by Corydon76 (license 14) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106554 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-30Add the 'n' option to SpeechBackground, which has the application not answer therussell1-6/+25
channel if it has not already been answered. (closes SPD-51) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@101082 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-1/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-5/+0
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: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06"show application <foo>" changes for clarity.mmichelson1-21/+29
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89044 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Merged revisions 79334 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79334 | file | 2007-08-13 18:57:20 -0300 (Mon, 13 Aug 2007) | 2 lines Instead of accepting a single DTMF character accept a full string. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79335 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-13Merged revisions 79207 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r79207 | file | 2007-08-13 11:51:09 -0300 (Mon, 13 Aug 2007) | 2 lines Add an API call to allow the engine to know that DTMF was received. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79208 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵tilghman1-151/+153
few other formatting cleanups, too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-25Merged revisions 77176 via svnmerge from file1-2/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines (closes issue #10303) Reported by: jtodd Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77182 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-81/+12
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16It is no longer required for each module that deals with a channel to call ↵file1-2/+0
ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11Use the linkedlists.h AST_LIST_NEXT macro for modifying the list of results.file1-5/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74616 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-11Allow the native formats of a channel to influence the audio that is going ↵file1-2/+2
to the engine. The best format will try to be chosen with an ultimate fallback to signed linear if possible. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74570 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-22Merged revisions 71068 via svnmerge from qwell1-2/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r71068 | qwell | 2007-06-22 10:00:30 -0500 (Fri, 22 Jun 2007) | 12 lines Merged revisions 71065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r71065 | qwell | 2007-06-22 09:52:18 -0500 (Fri, 22 Jun 2007) | 4 lines Fix a few silly usages of ast_playstream() - it only ever returns 0... Issue 10035 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71069 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-15Merged revisions 69558 via svnmerge from file1-1/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r69558 | file | 2007-06-15 15:23:45 -0400 (Fri, 15 Jun 2007) | 2 lines Add support for setting the maximum length of acceptable DTMF in SpeechBackground. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69559 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-14Convert uses of strdup() to ast_strdup()russell1-3/+3
(issue #9983, eliel) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@69436 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06Issue 9869 - replace malloc and memset with ast_calloc, and other coding ↵tilghman1-1/+1
guidelines changes git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-13Merged revisions 61651 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61651 | file | 2007-04-13 14:08:02 -0400 (Fri, 13 Apr 2007) | 2 lines Do not bother looking for a result if none are present. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61652 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06Merged revisions 60361 via svnmerge from file1-15/+52
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60361 | file | 2007-04-05 22:14:00 -0300 (Thu, 05 Apr 2007) | 2 lines Add support for returning different types of results (ie: NBest). ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60362 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-03Merged revisions 59963 via svnmerge from file1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59963 | file | 2007-04-03 15:40:59 -0400 (Tue, 03 Apr 2007) | 2 lines Don't clash when a person both speaks and uses DTMF. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59969 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26Merged revisions 59223 via svnmerge from file1-3/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59223 | file | 2007-03-26 16:34:14 -0300 (Mon, 26 Mar 2007) | 2 lines Add ability to specify no timeout. This means as soon as the prompt is done playing it moves on to the next priority. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59224 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26Merged revisions 59213 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59213 | file | 2007-03-26 14:13:06 -0400 (Mon, 26 Mar 2007) | 2 lines Make SpeechBackground obey the digit timeout value. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59214 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Merged revisions 57053 via svnmerge from file1-3/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57053 | file | 2007-02-28 12:45:50 -0500 (Wed, 28 Feb 2007) | 2 lines Better handle timeouts when the individual speaks after everything has been played but before the timeout ends. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@57054 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-21Merged revisions 55947 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55947 | file | 2007-02-21 15:03:38 -0500 (Wed, 21 Feb 2007) | 2 lines Only start playing the next file if we have not been quieted. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@55948 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-16Merged revisions 54714 via svnmerge from file1-26/+30
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54714 | file | 2007-02-15 19:48:48 -0500 (Thu, 15 Feb 2007) | 2 lines Don't let dtmf leak over into the engine and let it skew the results... also give DTMF results priority. (issue #9014 reported by surftek) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54715 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-08Merged revisions 53601 via svnmerge from file1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53601 | file | 2007-02-08 12:54:32 -0500 (Thu, 08 Feb 2007) | 2 lines Fix timeout issue when utterance is longer then timeout itself. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53602 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-18Merged revisions 51251 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51251 | file | 2007-01-18 14:17:34 -0500 (Thu, 18 Jan 2007) | 2 lines Only start timeout once we reach the end of the files to play back. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51252 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-10Merged revisions 50433 via svnmerge from file1-28/+25
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r50433 | file | 2007-01-10 15:25:44 -0500 (Wed, 10 Jan 2007) | 2 lines Merge speech-multi branch which adds support for joining multiple sound files together to be played one after another in SpeechBackground. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@50434 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-05const-ify some more APIs, and fix rev 49710 from branch-1.4 in a better way herekpfleming1-5/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@49711 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-05As per ToDo list, I have made it so that Wait(), WaitExten(), Congestion(), ↵murf1-1/+1
Busy(), Read(), WaitForRing(), will now either actually handle a floating point argument as advertised, or has been upgraded to accept a floating point [timeout] arg. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44435 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-24Documentation updates (thanks Shaun for the speechrec.txt one!)file1-2/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40968 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21merge new_loader_completion branch, including (at least):kpfleming1-59/+45
- 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: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-18Expand speech API so that the developer can interact with the engine more ↵file1-0/+25
directly and use specific functions of the connector even if a generic API call is not available git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37881 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-11don't leak a frame when breaking out of the loop on a timeoutrussell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33448 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-4/+4
again :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-30fix various typos and other bits (from Ian Kinner)kpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@30800 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-24Nothing to see here... move alongfile1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@30131 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-24Update some documentation (file internal brain bug #42)file1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@30104 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-15Make sure that the channel is answered before doing SpeechBackground. (issue ↵file1-0/+6
#josh-wait-I-dont-have-issue-numbers) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@27194 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10remove almost all of the checks of the result from ast_strdupa() or alloca().russell1-4/+2
As it turns out, all of these checks were useless, because alloca will never return NULL. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26451 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-01remove \n from the end of a couple of synopsis fieldsrussell1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@23929 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-14This rather large commit changes the way modules are loaded. rizzo1-18/+6
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: http://svn.digium.com/svn/asterisk/trunk@20003 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-13Updates to speech recognition API and dialplan utilities. Moved to using ↵file1-184/+399
dialplan functions, and some other misc things. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@19645 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-12Add ability to see if the person calling said anything or not.file1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@19512 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-10Presenting a revised data stores and oh my, a generic speech recognition ↵file1-0/+575
API! I wonder what we can do with this now... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18979 f38db490-d61c-443f-a65b-d21fe96a405b
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