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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
2008-01-02note that chan_console requires portaudio v19Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-31Merge changes from team/russell/codec_resampleRussell Bryant
This commit imports libresample for use in Asterisk. It also adds a new codec module, codec_resample. This module uses libresample to re-sample signed linear audio between 8 kHz and 16 kHz. It also provides an alternative for converting between 16 kHz G.722 and 8 kHz signed linear when using G.722, which will likely be useful as some people have complained about volume issues when the current codec_g722 converts to 8 kHz signed linear. But, to test this, you will have to disable the g722-to-slin and g722-to-slin16 translators in codec_g722.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-31Merge the main set of changes from team/russell/chan_console.Russell Bryant
Add a new console channel driver, chan_console, which is a console channel driver that uses portaudio as a cross platform audio interface. It was written to provide a console channel driver that works with Mac CoreAudio, but it supports a number of other audio interfaces, as well, including OSS and ALSA. It could one day be the single console channel driver, but does not yet have as many features as chan_oss. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-28Some changes to app_amd. Mark Michelson
The channel name is printed in verbose messages maximumWordLength option added. Duration of words that do not meet the minimum word duration will be logged The duration of pre-greeting silence will be logged Only consider us in the greeting if we actually detected a valid word duration. (closes issue #11650, reported and patched by davevg) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-27clarify the type of video support in chan_ossLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-26Add a new dialplan function, DIALPLAN_EXISTS(), which allows you to check forRussell Bryant
the existence of a dialplan target. (closes issue #11579) Reported by: irroot Patches: func_dialplan2.c uploaded by irroot (license 52) -- Additional changes by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-26Adding support for storing the queue log entries in a realtime backend.Mark Michelson
(closes issue #11625, reported and patched by sergee) Thank you very much to sergee for adding this new feature! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-21The one documentation source I forgot to update after the merge of the ↵Mark Michelson
queue-penalty branch was the CHANGES file. No longer! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-19Reorganize CHANGES a bit. The "misc" section grew too large...Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-19Adding the ability to specify the To: header in an outbound INVITEOlle Johansson
by adding an exclamation mark to the dial string. This patch also exists for 1.4 in the fixtoheader-1.4 branch and has been in production for quite some time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-19Add option for starting remote Asterisk by naming the actual runtime socket ↵Olle Johansson
instead of pointing to configuration file with -C Reported by: sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax (license 359) doc changes by committer (closes issue #11598) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-16Adding a new CLI command for "manager reload", which is important now thatOlle Johansson
you need to reload after changes. Thanks YS. Reported by: ys Patches: trunk93163_manager_reload.c.diff uploaded by ys (license 281) (related to issue #11414) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-16Change manager so that registered accounts are stored in memory. This opens ↵Olle Johansson
for a manager realtime implementation. If you change accounts in manager.conf, you now need to reload to activate the changes (deletions, additions). This was not the case with 1.4. Reported by: ys Patches: trunk93163_manager_reload.c.diff uploaded by ys (license 281) (closes issue #11414) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-16Adding console_video to CHANGES. It's important that we keep this file up to ↵Olle Johansson
date, even with experimental stuff. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-16HUGE improvements to QoS/CoS handling by IgorGOlle Johansson
- Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-16Update documentationOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-14Remove use of privacy.conf by the Privacy app.Tilghman Lesher
Reported by: eliel Patch by: eliel (Closes issue #11344) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06Add manager command for showing all current channels.Olle Johansson
Thanks, eliel, for writing the original patch. Modified by me to follow other manager events and the new "moremanager" style. (closes issue #11478) Reported by: eliel Patches: manager.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-05Change cdr_manager to use a "CDR" level, rather than the (overcrowded) ↵Tilghman Lesher
"call" level. (Closes issue #11015) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-05Added multiple name listing. (Closes issue #10413)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04Add manager action 'sipshowregistry'.Jason Parker
Closes issue #11464, patch by eliel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04Add support for monitoring MWI on FXO lines.Russell Bryant
This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor option enables MWI monitoring. When the MWI state on a line changes, then the script specified by mwimonitornotify will be executed for custom handling of the state change, similar to the externnotify option of voicemail.conf. Also, when the MWI state on an FXO line changes, an internal Asterisk event is generated to indicate the new state of the associated mailbox. That may, any module that cares about MWI information will get notified and can handle it just as if app_voicemail had sent this notification. (BE-253, original patch from markster, with some minor modifications by me to add comments, documentation, and internal event support) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04(closes issue #11422)Olle Johansson
Reported by: eliel Patches: core.show.hint.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04(closes issue #11462)Olle Johansson
Reported by: eliel Patches: CHANGES.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-03Add AGI commands for speech recognition. These mirror the dialplan ↵Joshua Colp
applications mostly but present the information in a nicer fashion. The SPEECH RECOGNIZE command for example will return the results instead of having to query the dialplan functions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-28Adding support for realtime music on hold. The following are the main points:Mark Michelson
1. When moh is started, we search first in memory to find the class. If we do not find it in memory, we search realtime instead. 2. When moh is restarted (as in, it had been started on this particular channel, stopped, and now we're starting it again), if using the "files" mode, then realtime will always be rechecked. If you are using other modes, however, we will simply reattach to the external running process which was playing moh earlier in the call. This is a necessary compromise so that we don't end up with too many background processes. 3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes, then moh classes found in realtime will be added to the in-memory list. This has the advantage of not requiring database lookups each time moh is started, but it has the disadvantage of not truly being realtime. I have tested this for functionality, and it passes. I also tested this under valgrind and there are no memory problems reported under typical use. Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker! (closes issue #11196, reported and patched by sergee) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26- Mark "concise" as deprecatedOlle Johansson
- Restructure other changes to UPGRADE.txt and CHANGES We're still looking for scripts that replace asterisk -rx "show shannels concise" by using the manager interface, but still produces the same output. Anyone? git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26Thanks to pnlarsson for noting the spelling error in the cli commands. Also, ↵Steve Murphy
added some verbage about the new algorithm to CHANGES. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-25- Deprecate "call-limit" in chan_sip. No other channel driver enforces ↵Olle Johansson
call-limits and we now have the groupcount system to implement call-limits in the dialplan. You can use the "setvar" option in realtime/sip.conf to set limits per device. - Implement "callcounter" as a new option to enable the call counting we need to report device status to queue, manager and SIP subscriptions. The call counter setting is now enabled in the code by setting the device call-limit to 999. When we remove the call limit, we can simply enable this with a boolean setting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21Change Read to set READSTATUS as an indication of the resultTilghman Lesher
Also, some cleanup to CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman (Closes issue #11004) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21Merge changes from team/russell/sla_trunk_moh ...Russell Bryant
* Added the ability to specify the music on hold class used to play into the conference when there is only one member and the M option is used. * Added the ability to specify a music on hold class to play instead of ringing for the SLATrunk application. (patched by me, and tested internally) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19Changed the "busy-level" option in sip.conf to "busylevel" to be more parallelMark Michelson
with the SIPPEER() argument of the same name. The deprecation procedure is not being used here since this is a trunk-only option. (closes issue #11307, reported by pj, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19Adding SYSINFO() dialplan function for retrieval of system informationMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19Update CHANGESOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-13Update the ParkedCall application to grab the first available parked call if noRussell Bryant
parked extension is provided as an argument. (closes issue #10803) Reported by: outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by outtolunc (license 237) - modified by me to work a bit differently ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89250 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-07Print out the channel name as a prefix to the "agi debug" output. This makesRussell Bryant
AGI debugging on busy systems much easier. (closes issue #10730) Reported by: junky Patches: agi_debug_chan.diff uploaded by junky (license 177) 20070923_10730.diff uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89074 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Added the ability to do "meetme concise" with the "meetme" CLI command.Russell Bryant
This extends the concise capabilities of this CLI command to include listing all conferences, instead of an addition to the other sub commands for the "meetme" command. (closes issue #11078) Reported by: jthomas Patches: meetme-concise.patch uploaded by jthomas (license 293) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Adding the queue strategy wrandomMark Michelson
(closes issue #10942, reported and patched by julianjm, documentation changes by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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