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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
2007-11-06Added the S() and L() options to the MeetMe application. These are prettyRussell Bryant
much identical to the S() and L() options to Dial(). They let you set timeouts for the conference, as well as have warning sounds played to let the caller know how much time is left, and when it is running out. (closes issue #8030) Reported by: areski Patches: meetme_timeout_timelimit_v2.patch uploaded by areski (license 29) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Change wording to that suggested by MasterYodaTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88653 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Merge the code from asterisk/team/group/chan_unistim:Russell Bryant
This introduces a new channel driver, chan_unistim, that supports the Unistim VoIP protocol for Nortel phones. The following models have been confirmed to work: i2002, i2004 and i2050. (closes issue #8864) Reported by: c_hans Patches: chan_unistim.patch uploaded by c (license 304) ustm_no_conf.diff uploaded by junky (license 177) Tested by: c_hans, dbowerman, math, junky, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Add a few bytes on LUATilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-26Forgot to update CHANGES when I committed the linear queue strategy.Mark Michelson
Thank you Russell, for pointing this out! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-17Document the changes made earlier today to meetmeTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-15Added support for reading the TOUCH_MONITOR_PREFIX channel variable.Russell Bryant
It allows you to configure a prefix for auto-monitor recordings. (closes issue #6353) Reported by: ivanfm Patches: asterisk_automon_v4.patch uploaded by ivanfm (original patch) - updated patch: 6353-touch_monitor_prefix.diff uploaded by qwell (license 4) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-09Note jitterbuffer support for chan_local in CHANGESRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-13Added the ability to pause and unpause members via the CLIMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-13Add setvar support to chan_zap. Just like you can in chan_sip and chan_iax2 ↵Joshua Colp
you can now use it with zaptel channels. (done while in Montreal at the Asterisk bootcamp!) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-11(closes issue #9433)Joshua Colp
Reported by: junky Patches: register_trying.diff.txt uploaded by jcmoore Disable sending 100 Trying on REGISTER attempts and make it an option. This has been signed off by oej. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-06Add EXTENSION_STATE() function that can retrieve the state of an extension thatRussell Bryant
has a hint. (closes issue #10635, adamgundy) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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