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2010-03-04Merged revisions 250730 via svnmerge from mmichelson1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r250730 | mmichelson | 2010-03-04 14:12:26 -0600 (Thu, 04 Mar 2010) | 3 lines Adjust XML for func_channel to indicate that rtpdest can take a "text" argument. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@250733 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Merged revisions 243482 via svnmerge from russell1-3/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r243482 | russell | 2010-01-27 11:32:07 -0600 (Wed, 27 Jan 2010) | 13 lines Fix the ability to specify an OSP token for an outbound IAX2 call. When this patch was originally submitted, the code allowed for the token to be set via a channel variable. I decided that a cleaner approach would be to integrate it into the CHANNEL() function. Unfortunately, that is not a suitable approach. It's not possible to get the value set on the channel soon enough using that method. So, go back to the simple channel variable method. (closes issue #16711) Reported by: homesick Patches: iax-svn.diff uploaded by homesick (license 91) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@243485 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@211580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-11Update CHANNEL(transfercapabilities) documentation.lmadsen1-0/+11
(closes issue #15073) Reported by: pkempgen Patches: 20090511__issue15073__trunk.diff.txt uploaded by tilghman (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@193612 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merge a large set of updates to the Asterisk indications API.russell1-3/+10
This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-13Merged revisions 168561 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168562 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-01Merge changes from team/group/appdocsxmlrussell1-81/+162
This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-15Add some more IAX2-specific information about the channel to the CHANNEL()tilghman1-0/+9
function and begin the transition from SIPCHANINFO() to just using CHANNEL(). (closes issue #12856) Reported by: mostyn Patches: iax_and_sip_channel_info.patch uploaded by mostyn (license 398) (with some additional cleanup by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-05This patch adds more detailed statistics for RTP channels, and provides an ↵bbryant1-0/+12
API call to access it, including maximums, minimums, standard deviatinos, and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function for any channel that uses RTP. (closes issue #10590) Reported by: gasparz Patches: chan_sip_c.diff uploaded by gasparz (license 219) rtp_c.diff uploaded by gasparz (license 219) rtp_h.diff uploaded by gasparz (license 219) audioqos-trunk.diff uploaded by snuffy (license 35) rtpqos-trunk-r119891.diff uploaded by sergee (license 138) Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120635 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-03Add a function, CHANNELS(), which retrieves a list of all active channels.tilghman1-4/+69
(closes issue #11330) Reported by: rain Patches: func_channel-channel_list_function.diff uploaded by rain (license 327) (with some additional changes by me, mostly to meet coding guidelines) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120230 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-21(closes issue #6113)jpeeler1-0/+5
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-18Context tracing for channelstilghman1-0/+24
(closes issue #11268) Reported by: moy Patches: chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103754 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19another bunch of include removals (errno.h and asterisk/logger.h)rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89425 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
2007-07-31Mostly cleanup of documentation to substitute the pipe with the comma, but a ↵tilghman1-5/+5
few other formatting cleanups, too. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-26Add rtpdest option to SIP CHANNEL() dialplan function to return the IP ↵file1-0/+5
address and port that RTP (be it audio/video/text) is going to. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71988 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20Merge changes from team/russell/iax2_osprussell1-0/+3
This set of changes adds OSP support to chan_iax2. However, I have modified the patch a bit from what was submitted. You now use the CHANNEL() function to get and set the OSP token for IAX2. (issue #8531, reported by and original patch by homesick, patch updated by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61702 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-27Merged revisions 59256 via svnmerge from russell1-12/+31
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines Convert the RTPQOS function to just be additional parameter of the CHANNEL function. This way, it will be possible for other RTP based channel drivers to expose this information in the future. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@59257 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-24Doxygen updateoej1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51920 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-06finish const-ifying ast_func_read()kpfleming1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@49741 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-25rename the structs struct tone_zone_sound and struct tone_zonerizzo1-1/+1
defined in indications.h to ind_tone_zone_sound and ind_tone_zone, to avoid conflicts with the structs with the same names defined in tonezone.h Hope i haven't missed any instance. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48958 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-16Merged revisions 44809 via svnmerge from pcadach1-3/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44809 | pcadach | 2006-10-10 23:44:54 +0700 (Втр, 10 Окт 2006) | 1 line CHANNEL() function sometime mix parameter and value ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47718 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21merge new_loader_completion branch, including (at least):kpfleming1-15/+3
- 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
2006-06-07simplify autoconfig include mechanism (make tholo happy he can use lint ↵kpfleming1-3/+5
again :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-04Make tonezone writeable in CHANNEL() (from my old func_tonezone.c)oej1-2/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32018 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-25- mark some applications deprecated that already have replacementsrussell1-1/+17
- add BLACKLIST and mark LookupBlacklist deprecated - add transfercapability support to CHANNEL and mark SetTransferCapability deprecated (issue #7225, Corydon) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@30359 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10use the channel lock wrappers (issue #7120, Mithraen)russell1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26528 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-03 Provide the ability to adjust txgain/rxgain on a channel level via the ↵bweschke1-1/+11
CHANNEL() function git-svn-id: http://svn.digium.com/svn/asterisk/trunk@24621 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-14This rather large commit changes the way modules are loaded. rizzo1-9/+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
2006-04-08since the module API is changing, it's a good time to const-ify the ↵kpfleming1-2/+2
description() and key() return values git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18552 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-25Bug 6670 - Additional parameters to the CHANNEL functilghman1-1/+13
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@14870 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-21set keywords property correctlykpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10627 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-12major dialplan functions updatekpfleming1-0/+152
deprecate LANGUAGE() and MUSICCLASS(), in favor of CHANNEL() git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9674 f38db490-d61c-443f-a65b-d21fe96a405b
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