265 Commits (8ba94bb2587fd4c3a31e66b81594e04bba94925e)

Author	SHA1	Message	Date
Tilghman Lesher	caab83a81b	Merged revisions 119296 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r119296 | tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines Add native AGI command GOSUB, as invoking Gosub with EXEC does not work properly. (closes issue #12760) Reported by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	ad57cc89de	Merged revisions 118647 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines Merged revisions 118646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@118648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	085ef90375	Merged revisions 110881 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar 2008) | 18 lines Merged revisions 110880 via svnmerge from 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/branches/1.6.0@110882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	6aa9c04ee9	Merged revisions 110499 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines Note that the TCP and TLS support is currently considered experimental and is subject to change while we work out the remaining issues. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@110501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	764077c366	change the versions in the heading ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@105600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	ebcefd1395	Add a "devstate change" CLI command to control custom device states. Also, ... 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	18 years ago
Joshua Colp	2a7eac9940	Add an 'e' option to ResetCDR which re-enables a CDR that has been disabled. ... (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	18 years ago
Russell Bryant	86e26793c2	Update CHANGES for SMDI stuff ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	f274f7bcaa	Permit additional CDR columns to be saved in Postgres. Note that these ... 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	18 years ago
Tilghman Lesher	f92a3e119e	Move Originate to a separate privilege and require the additional System privilege to call out to a subshell. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	3e0f3915a5	Add CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan 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	18 years ago
Olle Johansson	17c761c5ff	- No space in manager event names, please ... - Add new event to CHANGES git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	26755e3882	Context tracing for channels ... (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	18 years ago
Mark Michelson	c08a40fb61	Document GotoIfTime change from svn revision 103738 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jeff Peeler	16a14a4cd8	Requested changes from Pari, reviewed by Russell. ... 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	18 years ago
Russell Bryant	2dd50b7656	remove entry that is no longer in the tree ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101373 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	0ca3d5509e	Update CHANGES with rtppage ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	46f06a5e0c	Fix a typo ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	22fae48e3c	Add the 'n' option to SpeechBackground, which has the application not answer the ... 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	18 years ago
Joshua Colp	3bf7daa0c0	Merge in strictrtp branch. This adds a strictrtp option to rtp.conf which 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	18 years ago
Jason Parker	3bd33214b9	Move code from res_features into (new file) main/features.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	cfa0ec1f97	Add res_config_ldap for realtime LDAP engine. ... (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	18 years ago
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