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2008-01-18Merge changes from team/group/sip-tcptlsrussell1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@99085 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-17Add support for an easy way to automatically execute some Asterisk CLI commandsrussell1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@98986 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16Info about res_config_curltilghman1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98984 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-16Add note about new update.log to CHANGES, by request of jmls and further ↵qwell1-0/+1
prodding by jsmith. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98969 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-14Add backupdeleted option to app_voicemailqwell1-0/+2
(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: http://svn.digium.com/svn/asterisk/trunk@98889 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
2008-01-13- 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/.russell1-2/+14
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
2008-01-12Add a new CLI command, "core set chanvar", which allows you to set a channelrussell1-0/+1
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
2008-01-12Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always ↵kpfleming1-0/+3
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
2008-01-11Add 'auto' signalling mode for Zaptel channels.kpfleming1-0/+7
(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
2008-01-11Add a new global and per-peer option to chan_sip, qualifyfreq, which allows yourussell1-0/+2
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
2008-01-10Several manager changes:tilghman1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@97651 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-09Added a new module, res_phoneprov, which allows auto-provisioning of phonestwilson1-0/+8
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: http://svn.digium.com/svn/asterisk/trunk@97634 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-08Adding the option of specifying a second interface in a member definition ↵mmichelson1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@97203 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-02note that chan_console requires portaudio v19kpfleming1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95839 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31Merge changes from team/russell/codec_resamplerussell1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@95501 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-31Merge the main set of changes from team/russell/chan_console.russell1-9/+13
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: http://svn.digium.com/svn/asterisk/trunk@95412 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-28Some changes to app_amd. mmichelson1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@95167 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-27clarify the type of video support in chan_ossrizzo1-1/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94902 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Add a new dialplan function, DIALPLAN_EXISTS(), which allows you to check forrussell1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@94799 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-26Adding support for storing the queue log entries in a realtime backend.mmichelson1-0/+1
(closes issue #11625, reported and patched by sergee) Thank you very much to sergee for adding this new feature! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94782 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-21The one documentation source I forgot to update after the merge of the ↵mmichelson1-0/+3
queue-penalty branch was the CHANGES file. No longer! git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94546 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Reorganize CHANGES a bit. The "misc" section grew too large...oej1-38/+53
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93899 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Adding the ability to specify the To: header in an outbound INVITEoej1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@93897 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-19Add option for starting remote Asterisk by naming the actual runtime socket ↵oej1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@93854 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16Adding a new CLI command for "manager reload", which is important now thatoej1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@93166 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16Change manager so that registered accounts are stored in memory. This opens ↵oej1-4/+9
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: http://svn.digium.com/svn/asterisk/trunk@93165 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16Adding console_video to CHANGES. It's important that we keep this file up to ↵oej1-0/+4
date, even with experimental stuff. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93164 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16HUGE improvements to QoS/CoS handling by IgorGoej1-0/+5
- 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: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16Update documentationoej1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93160 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Remove use of privacy.conf by the Privacy app.tilghman1-0/+2
Reported by: eliel Patch by: eliel (Closes issue #11344) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93066 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-06Add manager command for showing all current channels.oej1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@91347 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Change cdr_manager to use a "CDR" level, rather than the (overcrowded) ↵tilghman1-0/+2
"call" level. (Closes issue #11015) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91173 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Added multiple name listing. (Closes issue #10413)tilghman1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91172 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Add manager action 'sipshowregistry'.qwell1-0/+1
Closes issue #11464, patch by eliel. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90991 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Add support for monitoring MWI on FXO lines.russell1-1/+6
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: http://svn.digium.com/svn/asterisk/trunk@90949 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04(closes issue #11422)oej1-0/+1
Reported by: eliel Patches: core.show.hint.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90853 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04(closes issue #11462)oej1-0/+2
Reported by: eliel Patches: CHANGES.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90852 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03Add AGI commands for speech recognition. These mirror the dialplan ↵file1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@90656 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-28Adding support for realtime music on hold. The following are the main points:mmichelson1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@89946 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26- Mark "concise" as deprecatedoej1-3/+0
- 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: http://svn.digium.com/svn/asterisk/trunk@89606 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26Thanks to pnlarsson for noting the spelling error in the cli commands. Also, ↵murf1-0/+10
added some verbage about the new algorithm to CHANGES. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89583 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-25- Deprecate "call-limit" in chan_sip. No other channel driver enforces ↵oej1-1/+7
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: http://svn.digium.com/svn/asterisk/trunk@89554 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21Change Read to set READSTATUS as an indication of the resulttilghman1-25/+36
Also, some cleanup to CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman (Closes issue #11004) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89489 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21Merge changes from team/russell/sla_trunk_moh ...russell1-0/+4
* 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: http://svn.digium.com/svn/asterisk/trunk@89470 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19Changed the "busy-level" option in sip.conf to "busylevel" to be more parallelmmichelson1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@89441 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19Adding SYSINFO() dialplan function for retrieval of system informationmmichelson1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89421 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19Update CHANGESoej1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89407 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13Update the ParkedCall application to grab the first available parked call if norussell1-0/+3
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: http://svn.digium.com/svn/asterisk/trunk@89250 f38db490-d61c-443f-a65b-d21fe96a405b
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