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2007-04-11added HAVE_SYSINFO preprocessor directives for portability and general happinessDwayne M. Hubbard
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11Add a configure script check for sysinfo support.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11added option_minmemfree for use in asterisk.conf to specify the amount of ↵Dwayne M. Hubbard
minimum free memory prior to accepting calls. added CLI 'core show sysinfo' to display system information git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11via 8118, a RealTime upgrade to make RT a complete storage abstraction. The ↵Steve Murphy
store/destroy mechanisms needed these missing peices. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10Issue 6082 - New DTMF event for managerTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61324 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10Add an option to the dial API for playing music instead of ringing to the ↵Russell Bryant
caller. I started this for use with SLA but ended up deciding not to use it. However, there is no reason not to put this part in, anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10Merged revisions 60989 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. This also adds the mods from 1.4/r.61136; ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-09Merged revisions 60850 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60850 | tilghman | 2007-04-08 22:01:12 -0500 (Sun, 08 Apr 2007) | 10 lines Merged revisions 60849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-06Merged revisions 60603 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines To be able to achieve the things that we would like to achieve with the Asterisk GUI project, we need a fully functional HTTP interface with access to the Asterisk manager interface. One of the things that was intended to be a part of this system, but was never actually implemented, was the ability for the GUI to be able to upload files to Asterisk. So, this commit adds this in the most minimally invasive way that we could come up with. A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in the parser, and updated it to be thread-safe. The ability to check permissions of active manager sessions was added by Dwayne Hubbard. Then, hacking this all together and do doing the modifications necessary to the HTTP interface was done by me. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-06Major res_speech cleanup. It looks much better now!Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-06Merged revisions 60361 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@60362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-30Merged revisions 59522 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1 line several changes via kpflemings review ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-30Merged revisions 59486 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1 line These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated from transfer situations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-27Enhancement via 8118: Realtime API extension: add methods store_func and ↵Steve Murphy
destroy_func, to make Realtime a complete database abstraction git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26Merged revisions 59207 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-26Merged revisions 59202 via svnmerge from Nadi Sarrar
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines * mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it. * add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in' (the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-20The fix for the AEL <<security hole>> (bug 9316) is here...Steve Murphy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-15Merged revisions 58947 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58947 | russell | 2007-03-15 18:53:26 -0500 (Thu, 15 Mar 2007) | 3 lines Add configure script checking for GTK2 and some additional Makefile targets to support gmenuselect ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-13Merge changes from team/russell/sqlite:Russell Bryant
* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a SQLite3 database. (issue #7149, alerios) * Add new module, res_config_sqlite, which adds realtime database configuration support for SQLite version 2. I decided that this was ok since we didn't have any realtime support for version 3. If someone ports this to version 3, then version 2 support can be removed or marked deprecated. (issue #7790, rbarun_proformatique) * Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom. Also, note that there were other modules on the bug tracker that did not make the cut because they provided some duplicated functionality. Those are: * cdr_sqlite3 (issue #6754, moy) * cdr_sqlite3 (issue #8694, bsd) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-06Add some documentation on the arguments to the base64 encode/decode functions.Russell Bryant
(inspired by issue #9215) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-03Expand datastores to add the notion of inheritance. This will be needed forTilghman Lesher
the conversion of IAX2 variables from the current custom method to ast_storage. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57691 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-01Merged revisions 57364 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines Merge changes from svn/asterisk/team/russell/sla_updates * Originally, I put in the documentation that only Zap interfaces would be supported on the trunk side. However, after a discussion with Qwell, we came up with a way to make IP trunks work as well, using some things already in Asterisk. So, here it is, this now officially supports IP trunks. * Update the SLA documentation to reflect how to setup IP trunks. * Add a section in sla.txt that describes how to set up an SLA system with voicemail. * Simplify the way DTMF passthrough is handled in MeetMe. * Fix a bug that exposed itself when using a Local channel on the trunk side in SLA. The station's channel needs to be passed to the dial API when dialing the trunk. * Change a WARNING message to DEBUG in channel.h. This message is of no use to users. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-28Convert the PBX core to use read/write locks. This yields a nifty ↵Joshua Colp
performance improvement when it comes to simultaneous calls going through the dialplan. Using murf's test the old mutex based core took an average of 57.3 seconds while the rwlock based core took 31.1 seconds. That's a nifty 26.2 seconds performance improvement. The other good part is that if we do need to switch back then we just have to change the lock/unlock API calls. I converted everywhere that used to touch the mutex locks directly to use them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@57184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Doxygen additions, correctionsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Doxygen updates and correctionsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Creating new doxygen macro "\extref" to create page that lists Olle Johansson
external libraries and URLs to these. Please help me add these references. We might want to create a similar macro "\linuxpackage" to list the needed Linux packages in popular distributions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Add some external referencesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24Doxygen updates for AJI - The Asterisk Jabber APIOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22move the ast_module_info structure into the special section as well, ↵Kevin P. Fleming
otherwise when restore_globals() is called it will lose its pointer to the ast_module structure that the loader put there git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22give embedded modules a helping hand by backing up and restoring their ↵Kevin P. Fleming
global variables when they are loaded and unloaded git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-20Merged revisions 55590 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55590 | russell | 2007-02-20 13:57:07 -0600 (Tue, 20 Feb 2007) | 2 lines Increase the maximum number of manager headers to 128, at the request of Pari. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-17Merged revisions 55052 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r55052 | russell | 2007-02-16 18:40:34 -0600 (Fri, 16 Feb 2007) | 3 lines If the pg_config application is found, but there is probably executing it, then consider postgres unavailable. (issue #8637) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16Adding Realtime Text support (T.140) to AsteriskOlle Johansson
T.140/RFC 2793 is a live communication channel, originally created for IP based text phones for hearing impaired. Feels very much like the old Unix talk application. This code is developed and disclaimed by John Martin of Aupix, UK. Tested for interoperability by myself and Omnitor in Sweden, the company that wrote most of the specifications. A big thank you to everyone involved in this. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-14New CLI command "Core show settings" to list some core settingsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-13This introduces a new dialplan function, DEVSTATE, which allows you to do someRussell Bryant
pretty cool things. First, you can get the device state of anything in the dialplan: NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)}) NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)}) Most importantly, this allows you to create custom device states so you can control phone lamps directly from the dialplan. Set(DEVSTATE(Custom:mycustomlamp)=BUSY) ... exten => mycustomlamp,hint,Custom:mycustomlamp git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-13Merged revisions 54218 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54218 | russell | 2007-02-13 14:56:50 -0600 (Tue, 13 Feb 2007) | 3 lines Fix the documentation on the return values from device state provider registration and deletion. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-13- Constify the format string passed to ast_cli()Russell Bryant
- Simplify printing out the warranty and license git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12Merged revisions 54103 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) | 2 lines Change ast_set_state_callback() to ast_dial_set_state_callback() ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12Merged revisions 54066 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines - Add the ability to register a callback to monitor state changes in an asynchronous dial operation. - Rename the various references to "status" to "state" in the dial API ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10Merged revisions 53810 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines Merge team/russell/sla_rewrite This is a completely new implementation of the SLA functionality introduced in Asterisk 1.4. It is now functional and ready for testing. However, I will be adding some additional features over the next week, as well. For information on how to set this up, see configs/sla.conf.sample and doc/sla.txt. In addition to the changes in app_meetme.c for the SLA implementation itself, this merge brings in various other changes: chan_sip: - Add the ability to indicate HOLD state in NOTIFY messages. - Queue HOLD and UNHOLD control frames even if the channel is not bridged to another channel. linkedlists.h: - Add support for rwlock based linked lists. dial.c: - Add the ability to run ast_dial_start() without a reference channel to inherit information from. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-30Merged revisions 52997 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52997 | russell | 2007-01-30 17:23:24 -0600 (Tue, 30 Jan 2007) | 5 lines When we are checking for a system installed version of libgsm, we need to check for gsm.h as well. Furthermore, when checking for this header, it may be located in a gsm/ sub directory, so check for that, as well. (issue #8773) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-29Merged revisions 52494,52506 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52494 | jdixon | 2007-01-28 22:18:36 -0600 (Sun, 28 Jan 2007) | 4 lines Fixed problem with jitterbuf, whereas it would not complain about, and would allow itself to be overfilled (per the max_jitterbuf parameter). Now it rejects any data over and above that size, and complains about it. ........ r52506 | russell | 2007-01-29 10:54:27 -0600 (Mon, 29 Jan 2007) | 5 lines Clean up a few things in the last commit to the adaptive jitterbuffer code. - Specifically indicate to the compiler that the "dropem" variable only needs one but. - Change formatting to conform to coding guidelines. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-24Merged revisions 52107 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52107 | russell | 2007-01-24 15:42:47 -0600 (Wed, 24 Jan 2007) | 3 lines Fix the formatting of doxygen comments to properly indicate that the comment documents the previous entity, as opposed to the next one. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-24Merged revisions 52049 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r52049 | file | 2007-01-24 13:20:05 -0500 (Wed, 24 Jan 2007) | 2 lines Merge in dialing API and the app_page that uses it. (issue #BE-118) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@52050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-24Doxygen updatesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-20Add a comment that the frame type constants are transmitted directly over IAX2.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19Merged revisions 51311 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines Merge the changes from the /team/group/vldtmf_fixup branch. The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19As the comment in the diff says:Luigi Rizzo
AST_INLINE_API() is a macro that takes a block of code as an argument. Using preprocessor #directives in the argument is not supported by all compilers, and it is a bit of an obfuscation anyways, so avoid it. As a workaround, define a macro that produces either its argument or nothing, and use that instead of #ifdef/#endif within the argument to AST_INLINE_API(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19Regenerate configure script to reflect recent zaptel changesRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19Include tonezone.h for linux, tooRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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