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2007-05-18Merged revisions 65200 via svnmerge from murf1-2/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines Merged revisions 65172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65202 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-18Issue #5930 - Remove dependencies on res_adsi.so - clwadeoej1-30/+31
A big THANK YOU to clwade for this patch. Minor modifications by me. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64921 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-17Merged revisions 64820 via svnmerge from tilghman1-6/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64820 | tilghman | 2007-05-17 16:19:34 -0500 (Thu, 17 May 2007) | 10 lines Merged revisions 64819 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64819 | tilghman | 2007-05-17 16:14:36 -0500 (Thu, 17 May 2007) | 2 lines How is it that we never caught that this is returning the opposite of our documentation, until now? ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64821 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-15Add two new dialplan functions: ENUMQUERY and ENUMRESULT. These functionsrussell1-1/+29
allow you to initiate an ENUM query using ENUMQUERY, and then access the details of all of the results using ENUMRESULT. Previously, if you wanted to access multiple results, Asterisk would have to do a new DNS lookup every time. (patch by bbryant) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64480 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-08I noted this on the dev list but got no response, so I just did it myself.russell1-1/+4
Lock the call features when being used in chan_sip. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63447 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-07Merged revisions 63286 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63286 | file | 2007-05-07 17:45:01 -0400 (Mon, 07 May 2007) | 10 lines Merged revisions 63285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 lines Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63287 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-07Constificationsoej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63240 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-04a small upgrade to the coding standard, and an update to the code that ↵murf1-2/+2
triggered the upgrade. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63048 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-04Added a small bit of code to support the SNOM 360's Record button. Made the ↵murf1-0/+4
find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63046 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-04- Add manager command CoreSettingsoej3-0/+4
- Add missing option to options.h - Add missing variables to asterisk.h - Move manager version to manager.h include file git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63030 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30Add support for setting the CoS for VLAN traffic (802.1p) in Linux. Therussell4-4/+9
file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62457 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30Merged revisions 62414 via svnmerge from russell1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62414 | russell | 2007-04-30 10:25:31 -0500 (Mon, 30 Apr 2007) | 4 lines When serving dynamic content, include a Cache-Control header to instruct the browsers to not store the resulting content. (issue #9621, reported by Pari, patch by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62415 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-28Merge changes from team/russell/eventsrussell3-0/+537
This set of changes introduces a new generic event API for use within Asterisk. I am still working on a way for events to be shared between servers, but this part is ready and can already be used inside of Asterisk. This set of changes introduces the first use of the API, as well. I have restructured the way that MWI (message waiting indication) is handled. It is now event based instead of polling based. For example, if there are a bunch of SIP phones subscribed to mailboxes, then chan_sip will not have to constantly poll the mailboxes for changes. app_voicemail will generate events when changes occur. See UPGRADE.txt and CHANGES for some more information on the effects of these changes from the user perspective. For developer information, see the text in include/asterisk/event.h. As always, additional feedback is welcome on the asterisk-dev mailing list. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62292 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-28Remove a message that goes to LOG_ERROR that's not really an error.russell1-3/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62264 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25Merged revisions 61805 via svnmerge from file1-1/+18
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61806 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24Merged revisions 61781 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines Improve DTMF handling in ast_read() even more in response to a discussion on the asterisk-dev mailing list. I changed the enforced minimum length of a digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in between digits. These values are not configurable in a configuration file right now, but they can be easily changed near the top of main/channel.c. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61782 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20Merged revisions 61690 via svnmerge from russell1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61690 | russell | 2007-04-20 13:19:18 -0500 (Fri, 20 Apr 2007) | 4 lines Fix the UpdateConfig manager action to properly treat "variables" and "objects" differently (a=b versus a=>b). (issue #9568, reported by pari, patch by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61691 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-16Doxygen changesoej1-9/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61667 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11changed #if HAVE_SYSINFO to #if defined(HAVE_SYSINFO)dhubbard1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61576 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11added HAVE_SYSINFO preprocessor directives for portability and general happinessdhubbard1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61575 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11Add a configure script check for sysinfo support.file1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61557 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11added option_minmemfree for use in asterisk.conf to specify the amount of ↵dhubbard1-0/+1
minimum free memory prior to accepting calls. added CLI 'core show sysinfo' to display system information git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61539 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-11via 8118, a RealTime upgrade to make RT a complete storage abstraction. The ↵murf1-0/+17
store/destroy mechanisms needed these missing peices. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61374 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-10Issue 6082 - New DTMF event for managertilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61324 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-10Add an option to the dial API for playing music instead of ringing to the ↵russell1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@61259 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-10Merged revisions 60989 via svnmerge from murf1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@61152 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-09Merged revisions 60850 via svnmerge from tilghman1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@60851 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06Merged revisions 60603 via svnmerge from russell3-0/+30
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: http://svn.digium.com/svn/asterisk/trunk@60604 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06Major res_speech cleanup. It looks much better now!file1-6/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@60363 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-06Merged revisions 60361 via svnmerge from file1-0/+13
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
2007-03-30Merged revisions 59522 via svnmerge from murf1-7/+7
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: http://svn.digium.com/svn/asterisk/trunk@59523 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-30Merged revisions 59486 via svnmerge from murf1-0/+12
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: http://svn.digium.com/svn/asterisk/trunk@59500 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-27Enhancement via 8118: Realtime API extension: add methods store_func and ↵murf1-0/+4
destroy_func, to make Realtime a complete database abstraction git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59253 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26Merged revisions 59207 via svnmerge from russell1-2/+13
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: http://svn.digium.com/svn/asterisk/trunk@59208 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26Merged revisions 59202 via svnmerge from nadi1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@59203 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-20The fix for the AEL <<security hole>> (bug 9316) is here...murf1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59073 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-15Merged revisions 58947 via svnmerge from russell1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@58948 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-13Merge changes from team/russell/sqlite:russell1-0/+6
* 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: http://svn.digium.com/svn/asterisk/trunk@58866 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-06Add some documentation on the arguments to the base64 encode/decode functions.russell1-0/+19
(inspired by issue #9215) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@58149 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-03Expand datastores to add the notion of inheritance. This will be needed fortilghman1-0/+7
the conversion of IAX2 variables from the current custom method to ast_storage. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@57691 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-01Merged revisions 57364 via svnmerge from russell1-9/+10
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: http://svn.digium.com/svn/asterisk/trunk@57365 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-28Convert the PBX core to use read/write locks. This yields a nifty ↵file1-4/+22
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: http://svn.digium.com/svn/asterisk/trunk@57184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24Doxygen additions, correctionsoej7-14/+56
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56665 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24Doxygen updates and correctionsoej2-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56648 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24Creating new doxygen macro "\extref" to create page that lists oej1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@56647 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24Add some external referencesoej1-1/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56629 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-24Doxygen updates for AJI - The Asterisk Jabber APIoej2-0/+26
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56623 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22move the ast_module_info structure into the special section as well, ↵kpfleming1-1/+3
otherwise when restore_globals() is called it will lose its pointer to the ast_module structure that the loader put there git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56209 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-22give embedded modules a helping hand by backing up and restoring their ↵kpfleming1-0/+73
global variables when they are loaded and unloaded git-svn-id: http://svn.digium.com/svn/asterisk/trunk@56092 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-20Merged revisions 55590 via svnmerge from russell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@55591 f38db490-d61c-443f-a65b-d21fe96a405b
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