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2008-01-14Make sure the user's manager secret exists, even if it is blank.file1-1/+1
(closes issue #11749) Reported by: srt git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98830 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Add description of TOUPPER and TOLOWER dialplan functions to CHANGES.twilson1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98811 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Add proper call forwarding (all and busy) support for chan_skinny.qwell1-58/+315
Note: NoAnswer support is currently not implemented, as it would take a significant amount of work to figure out how to do correctly. Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98776 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Merged revisions 98774 via svnmerge from russell1-4/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | 3 lines Revert a change that introduces an unacceptable performance hit and is causing memory leaks ... (from rev 97973) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98775 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Fix for potential crash with vmextenqwell1-2/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98773 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Merged revisions 98737 via svnmerge from mmichelson1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan 2008) | 3 lines Fixing another compilation error. I'm a bit off today :( ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98738 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Blocked revisions 98734 via svnmergemmichelson0-0/+0
........ r98734 | mmichelson | 2008-01-14 10:30:33 -0600 (Mon, 14 Jan 2008) | 3 lines Oops. Last commit had compilation error. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98736 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Merged revisions 98733 via svnmerge from mmichelson1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan 2008) | 8 lines Adding explicit defaults for missing options to init_queue. This is necessary because if a user either removes or comments one of these options and reloads their queues, the option will not reset to its default, instead maintaining the value from prior to the reload. Thanks to John Bigelow for pointing this error out to me. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98735 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Print out a warning when spaces are used in the variable name in Set and ↵file1-2/+7
MSet. It is extremely hard to debug this issue so this should make it easier. (closes issue #11759) Reported by: caio1982 Patches: setvar_space_warning1.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98714 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Update documentation.file3-4/+7
(closes issue #11763) Reported by: IgorG Patches: docupd.v1.diff uploaded by IgorG (license 20) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98695 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Add another small option for the JACK app and JACK_HOOK function. The 'n'russell1-6/+18
option tells JACK not to start jackd automatically if it is not already running. Otherwise, the default is that jackd will get started for you if it isn't running already. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98676 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-13- Break up the Misc. section a bit with a new section for Misc. New Modulesrussell1-68/+72
- Change spacing a bit in some places for consistent indentation git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98656 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-13Bring in the code from team/russell/jack/.russell2-2/+973
Add a new module, app_jack, which provides interfaces to JACK, the Jack Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are provided; there is a JACK() application, and a JACK_HOOK() function. Both interfaces create an input and output JACK port. The application makes these ports the endpoint of the call. The audio coming from the channel goes out the output port and whatever comes back in on the input port is what gets sent to the channel. The JACK_HOOK() function turns on a JACK audiohook on the channel. This lets you run the audio coming from a channel through JACK, and whatever comes back in is what gets forwarded on as the channel's audio. This is very useful for building custom vocoders or doing recording or analysis of the channel's audio in another application. In case anyone is curious, the platform that inspired me to write this is PureData (http://puredata.info/). I wrote these JACK interfaces so that I could use Pd to do interesting things with the audio of phone calls ... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98628 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-13Add configure script check for JACK.russell5-23/+334
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98605 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-13Remove KDE configure script check that isn't usedrussell5-314/+11
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98604 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-13Remove a duplicate lock of the audiohook lock when destroying manipulaterussell1-1/+0
audiohooks. This causes an error when we attempt to destroy the lock later when freeing the audiohook. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98581 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-12Add a new CLI command, "core set chanvar", which allows you to set a channelrussell2-1/+41
variable (or function) on an active channel from the CLI. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98558 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-12Conversion to load manager.conf into memory did not convert the passwordtilghman1-3/+2
functions correctly. (Closes issue #11749) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98536 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-12merging a comment added in 1.4pari1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98514 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-12Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always ↵kpfleming2-10/+87
gets generated. (closes issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded by tzafrir (modified by me) (license 46) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98488 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-12Merged revisions 98467 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) | 4 lines Add a connection timeout attribute, as that was what was intended with the login timeout, but ODBC divides it up into 2 different timeouts. (Closes issue #11745) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98487 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Add some extra checking to help out with a potential error when trying torussell4-5/+59
run "make asterisk.pdf" when not all of the right packages are installed. (closes issue #10763) Reported by: Corydon76 Patches: 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98454 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Add 'auto' signalling mode for Zaptel channels.kpfleming3-47/+52
(closes issue #11690) Reported by: tzafrir Patches: signaling_to_signalling.diff uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir (license 46) zap_no_default_sig.diff uploaded by tzafrir (license 46) zap_signal_auto.diff uploaded by tzafrir (license 46) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98436 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Goodbye again drumkilla.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98435 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11drumkilla ftw.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98434 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11I am no longer Rockin'file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98432 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Testing something...file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98424 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Merged revisions 98390 via svnmerge from russell1-8/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) | 9 lines Fix up setting the EID on BSD based systems. (closes issue #11646) Reported by: caio1982 Patches: dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22) dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, mvanbaak ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98400 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Merged revisions 98325 via svnmerge from file1-0/+10
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines If the incoming RTP stream changes codec force the bridge to break if the other side does not support it. (closes issue #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch uploaded by tsearle (license 373) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98334 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Merged revisions 98317 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 lines If the channel is hungup during RECORD FILE send a result code of -1 to be uniform with everything else. (closes issue #11743) Reported by: davevg Patches: res_agi.diff uploaded by davevg (license 209) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98318 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Merged revisions 98315 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines Properly report the hangup cause as no answer when someone does not answer (closes issue #10574, reported by boch, patched by moy) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98316 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Kevin noted that the thing that I _actually_ changed here was that I convertedrussell1-3/+3
a value from a double, to a float, back to a double. Sure enough, when I changed my interim variable back to a double, it still blows up. Switching all of these to a float fixes the problem. This seems like a compiler bug where a double passed as an argument isn't getting properly aligned, so I'll have to see if I can replicate it with a small test program. (related to issue #11725) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98308 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Fix a bus error that happened when asterisk was built with optimizations on russell1-2/+6
with platforms that explode on unaligned access. I'm not exactly sure why this fixes it, but it fixed it on the machine I was testing on. If it makes sense to you, feel free to enlighten me. :) (closes issue #11725, patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98270 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Port Nick Gorham's timestamp patch to adaptive_odbc, tootilghman1-6/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98269 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Commit Nick Gorham's suggestion for timestamp fixtilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98268 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Blocked revisions 98265 via svnmergerussell0-0/+0
........ r98265 | russell | 2008-01-11 12:25:30 -0600 (Fri, 11 Jan 2008) | 11 lines Backport the ability to set the ToS bits on Linux when not running as root. Normally, we would not backport features into 1.4, but, I was convinced by the justification supplied by the supplier of this patch. He pointed out that this patch removes a requirement for running as root, thus reducing the potential impacts of security issues. (closes issue #11742) Reported by: paravoid Patches: libcap.diff uploaded by paravoid (license 200) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98267 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Merged revisions 98219 via svnmerge from file1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution. (closes issue #10327) Reported by: kkiely ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98220 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11At one point during working on this module, I had the lin/lin16 versions of therussell1-34/+2
framein callbacks different. However, they are now the same again, so remove the duplicate code and use the same functions for the lin/lin16 versions. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Merged revisions 98164 via svnmerge from tilghman1-143/+80
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) | 2 lines Back out changes from revision 97077, since it wasn't perfect ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98193 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Documentation updatestilghman1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98152 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Ascom phones send Flash events as SIP INFO using '!' as the 'digit'kpfleming1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98124 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Blocked revisions 98082 via svnmergerussell0-0/+0
........ r98082 | russell | 2008-01-10 21:39:33 -0600 (Thu, 10 Jan 2008) | 2 lines Fix samples vs. length calculations for g722 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98083 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11 - Fix the last set of places where incorrect assumptions were made about therussell2-7/+11
sample length with g722. It is _2_ samples per byte, not 1. This was all over the place, and I believed it, and it is what caused me to take so long to figure out what was broken. - Update copyright information on codec_g722. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98081 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Fix "core show translation" to not output information for "unknown" codecs.mmichelson1-1/+6
This fix was made in favor of the proposed patch since it doesn't involve changing a core codec define. (closes issue #11722, reported and initially patched by caio1982, final patch by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98047 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Add a new global and per-peer option to chan_sip, qualifyfreq, which allows yourussell3-4/+38
to set the qualify frequency. (closes issue #11597) Reported by: wilder Patches: qualifyfreq5.patch uploaded by wilder (license 362) -- with some mods by me git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98027 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Blocked revisions 98025 via svnmergerussell0-0/+0
........ r98025 | russell | 2008-01-10 18:14:59 -0600 (Thu, 10 Jan 2008) | 3 lines Simplify this code with a suggestion from Luigi on the asterisk-dev list. Instead of using is16kHz(), implement a format_rate() function. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98026 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-11Simplify this code with a suggestion from Luigi on the asterisk-dev list.russell1-19/+6
Instead of using is16kHz(), implement a format_rate() function. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98024 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Merged revisions 97973 via svnmerge from tilghman2-5/+15
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines 1) When we get a translated frame out, clone it, because if the translator pvt is freed before we use the frame, bad things happen. 2) Getting a failure from ast_sched_delete means that the schedule ID is currently running. Don't just ignore it. (Closes issue #11698) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97978 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Merged revisions 97976 via svnmerge from russell1-7/+37
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines Fix various timing calculations that made assumptions that the audio being processed was at a sample rate of 8 kHz. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97977 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Fix various issues in codec_g722.russell1-15/+50
- The most common fix being made here is to fix all of the places where the number of output samples and output bytes gets updated in the translator state structure. - Fix a number of other places where the number of samples provided as an initialization value to a struct was incorrect. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97975 f38db490-d61c-443f-a65b-d21fe96a405b
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