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2008-03-26Add a linkedlist macro that maintains a sorted listTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111036 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26Oops, fix this, tooTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-26Merged revisions 110880 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-25Add a special dialplan variable to chan_sip which will cause an audio file ↵Joshua Colp
to be played upon completion of an attended transfer. (closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-21Note that the TCP and TLS support is currently considered experimental andRussell Bryant
is subject to change while we work out the remaining issues. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-21Add note of the added Directory options, from commit 110237 (closes issue #7151)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19This change adds DNS manager support for registrations not referencing a ↵Jeff Peeler
peer entry. It looks like there is support for DNS manager for realtime peers as well, however it is not implemented correctly. The improper usage occurs when ast_dnsmgr_lookup is called with one of the arguments being an address from the stack to be continually updated. The variable from the stack will go out of scope and dnsmgr will continue to try and update the memory there, causing possible stack corruption. This problem will be worked on next as well as adding DNS manager support for peer entries. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-19Add the ability to use a pattern match for a hint.Joshua Colp
(closes issue #7767) Reported by: Corydon76 Patches: 20070314__simple_hint_lookup.diff.txt uploaded by Corydon76 pbx-trunk-98436.diff uploaded by plack (license 365) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18Add option 'randomperiodicannounce' to queues.conf. Setting this willMark Michelson
allow the list of periodic announcments specified to be played in a random order instead of being played sequentially. (closes issue #6681) Reported by: alt_phil Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-18Add manager peerstatus events when peer can't authenticate.Olle Johansson
(closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-13documenting changes as a result of adding TCP functionality to ExternalIVRJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-12add support for named sections in zapata.conf, and fix a few bugs in config ↵Kevin P. Fleming
file parsing (closes issue #9503) Reported by: tzafrir Patches: fix_cleanups uploaded by tzafrir (license 46) zapata_sections uploaded by tzafrir (license 46) skipchannel_options uploaded by tzafrir (license 46) conf_sample uploaded by tzafrir (license 46) patches updated by me to better conform to coding guidelines and fix some problems git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-11Add a trivial new dialplan function, AST_CONFIG(), which allows you to accessRussell Bryant
a variable from an Asterisk configuration file in the dialplan, or anywhere else where dialplan functions can be used. (Inspired by a discussion with Tilghman and Pari) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05Adding the Atxfer manager command. With this, you may initiateMark Michelson
an attended transfer over AMI (closes issue #10585) Reported by: ornati Patches: atxfer-trunk-r90428.diff uploaded by ornati (license 210) (with modifications from me) Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05Create a centralized configuration option for silencethresholdTilghman Lesher
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-04Update CHANGES headingRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105597 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-01Add a "devstate change" CLI command to control custom device states. Also,Russell Bryant
do some additional code cleanup and improvement in passing. (closes issue #12106) Reported by: nizon Patches: devstate-patch.txt uploaded by nizon (license 415) -- Updated to trunk, and tab completion added by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-26Add an 'e' option to ResetCDR which re-enables a CDR that has been disabled.Joshua Colp
(closes issue #11170) Reported by: kratzers Patches: ResetCDR.1.diff uploaded by kratzers (license 307) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-26Update CHANGES for SMDI stuffRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-25Permit additional CDR columns to be saved in Postgres. Note that theseTilghman Lesher
changes are backward-compatible, so no changes to UPGRADE.txt are necessary. (closes issue #9279) Reported by: rottenroddy Patches: 20080125__bug9279.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-22Move Originate to a separate privilege and require the additional System ↵Tilghman Lesher
privilege to call out to a subshell. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-19Add CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan ↵Joshua Colp
application. This will either be set to NOCHANNEL if the given channel was not found or SUCCESS if it worked. (closes issue #11553) Reported by: johan Patches: UPGRADE.txt.channelredirect.patch uploaded by johan (license 334) CHANGES.channelredirect.patch uploaded by johan (license 334) app_channelredirect-20080219.patch uploaded by johan (license 334) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-18- No space in manager event names, pleaseOlle Johansson
- Add new event to CHANGES git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-18Context tracing for channelsTilghman Lesher
(closes issue #11268) Reported by: moy Patches: chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-15Document GotoIfTime change from svn revision 103738Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-12Requested changes from Pari, reviewed by Russell.Jeff Peeler
Added ability to retrieve list of categories in a config file. Added ability to retrieve the content of a particular category. Added ability to empty a context. Created new action to create a new file. Updated delete action to allow deletion by line number with respect to category. Added new action insert to add new variable to category at specified line. Updated action newcat to allow new category to be inserted in file above another existing category. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-31remove entry that is no longer in the treeRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-30Update CHANGES with rtppageOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-30Fix a typoJason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-30Add the 'n' option to SpeechBackground, which has the application not answer theRussell Bryant
channel if it has not already been answered. (closes SPD-51) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-24Merge in strictrtp branch. This adds a strictrtp option to rtp.conf which ↵Joshua Colp
drops packets that do not come from the remote party. (closes issue #8952) Reported by: amorsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-23Move code from res_features into (new file) main/features.cJason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-22Add res_config_ldap for realtime LDAP engine.Tilghman Lesher
(closes issue #5768) Reported by: mguesdon Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121) res_ldap.conf.sample uploaded by suretec (license 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70) asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested by: oej, mguesdon, suretec, cthorner git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-22Documentation updates for BRIDGEPVTCALLIDOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-22Change the Asterisk CLI startup commands feature to read commands to run ↵Russell Bryant
from cli.conf after a discussion on the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-18Merge changes from team/group/sip-tcptlsRussell Bryant
This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-17Add support for an easy way to automatically execute some Asterisk CLI commandsRussell Bryant
immediately at startup. Any commands in the startup_commands file in the Asterisk config diretory will get executed. (closes issue #11781) Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176) -- With some changes by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-16Info about res_config_curlTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-16Add note about new update.log to CHANGES, by request of jmls and further ↵Jason Parker
prodding by jsmith. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-14Add backupdeleted option to app_voicemailJason Parker
(closes issue #10740) Reported by: ruffle Patches: app_voicemail.diff uploaded by ruffle (license 201) 10740-voicemail.diff uploaded by qwell (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak (license 7) Tested by: blitzrage, mvanbaak, qwell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-14Add description of TOUPPER and TOLOWER dialplan functions to CHANGES.Terry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-13- Break up the Misc. section a bit with a new section for Misc. New ModulesRussell Bryant
- Change spacing a bit in some places for consistent indentation git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-13Bring in the code from team/russell/jack/.Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@98628 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-12Add a new CLI command, "core set chanvar", which allows you to set a channelRussell Bryant
variable (or function) on an active channel from the CLI. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-12Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always ↵Kevin P. Fleming
gets generated. (closes issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded by tzafrir (modified by me) (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-11Add 'auto' signalling mode for Zaptel channels.Kevin P. Fleming
(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: https://origsvn.digium.com/svn/asterisk/trunk@98436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-11Add a new global and per-peer option to chan_sip, qualifyfreq, which allows youRussell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@98027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-10Several manager changes:Tilghman Lesher
1) Add the Dialplan class, for NewExten and VarSet events, which should cut down on the volume of traffic in the Call class. 2) Permit some commands to be run from multiple classes, such as allowing DBGet to be run from either the System or the Reporting class. 3) Heavily document each class in the sample config, as there were several that made no sense to be in the write= line, and two that made no sense to be in the read= line (since they controlled no permissions there). (Closes issue #10386) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-09Added a new module, res_phoneprov, which allows auto-provisioning of phonesTerry Wilson
based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-08Adding the option of specifying a second interface in a member definition ↵Mark Michelson
for a queue. app_queue will monitor this second device's state for the member, even though it actually calls the first interface. This ability has been added for statically defined queue members, realtime queue members, and dynamic queue members added through the CLI, dialplan, or manager. (closes issue #11603, reported by acidv) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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