243 Commits (b35f8d03580aef2deca980d93a1ab16d5d5ff04c)

Author	SHA1	Message	Date
Olle Johansson	b35f8d0358	Documentation updates for BRIDGEPVTCALLID ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	d1ba37f1c9	Change the Asterisk CLI startup commands feature to read commands to run 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	18 years ago
Russell Bryant	b995c78c31	Merge changes from team/group/sip-tcptls ... 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	18 years ago
Russell Bryant	8a5e93d766	Add support for an easy way to automatically execute some Asterisk CLI commands ... 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	18 years ago
Tilghman Lesher	bba20a8360	Info about res_config_curl ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98984 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	f35fca049a	Add note about new update.log to CHANGES, by request of jmls and further prodding by jsmith. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	b875d0df01	Add backupdeleted option to app_voicemail ... (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	18 years ago
Terry Wilson	9c1a8af01d	Add description of TOUPPER and TOLOWER dialplan functions to CHANGES. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	17ed33fc42	- Break up the Misc. section a bit with a new section for Misc. New Modules ... - 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	18 years ago
Russell Bryant	f32aec9f8f	Bring in the code from team/russell/jack/. ... 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	18 years ago
Russell Bryant	d0c89ab7ed	Add a new CLI command, "core set chanvar", which allows you to set a channel ... 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	18 years ago
Kevin P. Fleming	4b0a63ffa2	Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always 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	18 years ago
Kevin P. Fleming	138799091c	Add 'auto' signalling mode for Zaptel channels. ... (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	18 years ago
Russell Bryant	5c2beee6c3	Add a new global and per-peer option to chan_sip, qualifyfreq, which allows you ... 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	18 years ago
Tilghman Lesher	857e3412f4	Several manager changes: ... 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	18 years ago
Terry Wilson	3570ad103d	Added a new module, res_phoneprov, which allows auto-provisioning of phones ... 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	18 years ago
Mark Michelson	427f17fd9d	Adding the option of specifying a second interface in a member definition 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	18 years ago
Kevin P. Fleming	b4e80a1083	note that chan_console requires portaudio v19 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	21cb767db7	Merge changes from team/russell/codec_resample ... 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	18 years ago
Russell Bryant	4e99cc88e2	Merge the main set of changes from team/russell/chan_console. ... 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	18 years ago
Mark Michelson	d9e0bb0e84	Some changes to app_amd. ... 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	18 years ago
Luigi Rizzo	2145f6b8b8	clarify the type of video support in chan_oss ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	55e3cb32cd	Add a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for ... 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	18 years ago
Mark Michelson	00d848c94e	Adding support for storing the queue log entries in a realtime backend. ... (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	18 years ago
Mark Michelson	b6eab6d084	The one documentation source I forgot to update after the merge of the 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	18 years ago
Olle Johansson	241f271a99	Reorganize CHANGES a bit. The "misc" section grew too large... ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	1d6b192ce0	Adding the ability to specify the To: header in an outbound INVITE ... 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	18 years ago
Olle Johansson	489a648d5d	Add option for starting remote Asterisk by naming the actual runtime socket 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	18 years ago
Olle Johansson	c92dafd551	Adding a new CLI command for "manager reload", which is important now that ... 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	18 years ago
Olle Johansson	130fe4000a	Change manager so that registered accounts are stored in memory. This opens 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	18 years ago
Olle Johansson	df17bc73f0	Adding console_video to CHANGES. It's important that we keep this file up to date, ... even with experimental stuff. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	17afebc1a6	HUGE improvements to QoS/CoS handling by IgorG ... - 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	18 years ago
Olle Johansson	00647ff5f7	Update documentation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	70cd3d0037	Remove use of privacy.conf by the Privacy app. ... 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	18 years ago
Olle Johansson	5af2cf109e	Add manager command for showing all current channels. ... 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	18 years ago
Tilghman Lesher	ce2f670228	Change cdr_manager to use a "CDR" level, rather than the (overcrowded) "call" level. ... (Closes issue #11015) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	d226c1d637	Added multiple name listing. (Closes issue #10413) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91172 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	3f677a718a	Add manager action 'sipshowregistry'. ... Closes issue #11464, patch by eliel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	f15be28fb0	Add support for monitoring MWI on FXO lines. ... 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	18 years ago
Olle Johansson	25cbb792b9	(closes issue #11422) ... 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	18 years ago
Olle Johansson	d5c7e96526	(closes issue #11462) ...	18 years ago
Joshua Colp	8bfdea3160	Add AGI commands for speech recognition. These mirror the dialplan 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	18 years ago
Mark Michelson	a42259c3ff	Adding support for realtime music on hold. The following are the main points: ... 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	18 years ago
Olle Johansson	130a2051fa	- Mark "concise" as deprecated ... - 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	18 years ago
Steve Murphy	2ec4b57622	Thanks to pnlarsson for noting the spelling error in the cli commands. Also, 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	18 years ago
Olle Johansson	07cb09ad86	- Deprecate "call-limit" in chan_sip. No other channel driver enforces 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	18 years ago
Tilghman Lesher	1c295be7a0	Change Read to set READSTATUS as an indication of the result ... 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	18 years ago
Russell Bryant	6335b4b30d	Merge changes from team/russell/sla_trunk_moh ... ... * 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	18 years ago
Mark Michelson	fb3b4f4937	Changed the "busy-level" option in sip.conf to "busylevel" to be more parallel ... 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	18 years ago
Mark Michelson	67f044d42a	Adding SYSINFO() dialplan function for retrieval of system information ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago