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2011-08-22Add option for logging congested calls as CONGESTION instead of NO_ANSWER in CDRJonathan Rose
This patch adds a CDR option to cdr.conf that will allow CDR files to log calls ending with congestion in a way that is unique from other unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec Davis Patches: cdr_congestion.diff.txt (License #5546) patch uploaded by Alec Davis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-16Merged revisions 332101 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332101 | rmudgett | 2011-08-16 12:17:28 -0500 (Tue, 16 Aug 2011) | 140 lines Merged revisions 332100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 Multi-parkinglot directs calls to wrong parkinglot. JIRA ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 ParkedCall() with no extension should pickup first available call and does not. JIRA AST-576 Issues with parking lots * Removed searching for parking lots by extension. Parking lots can only be found by the parking lot name since parking lot access extensions and spaces are not guaranteed to be unique. * Added parking_lot_name option to the Park and ParkedCall applications. Updated documentation for Park and ParkedCall applications. * Add parkext_exclusive configuration option to make parking entry extensions specify which parking lot they access. (closes issue ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi Quezada (closes issue ASTERISK-17430) Reported by: Philippe Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA AST-624 'next' setting for findslot does nothing * Reimplemented since findslot feature option broken by -r114655. (closes issue ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett JIRA ASTERISK-15792 Dialplan continues execution after transfer to park. This happens for DTMF attended transfer, DTMF blind transfer, and DTMF one-touch-parking if the party initiating these features also initiated the call. * Fixed the return code from the affected builtin features when parking a call. (closes issue ASTERISK-15792) Reported by: Mat Murdock Tested by: rmudgett, twilson JIRA AST-607 The courtesytone is not playing to the expected call when picking up a parked call. This is mostly a documentation problem. However, the option is not reset to the default when features.conf is reloaded. * Updated features.conf.sample documentation for courtesytone and parkedplay options. * Reset the parkedplay option to default when features.conf is reloaded. JIRA AST-615 AMI Park action followed by features reload results in orphaned channels in parking lot. * Reloading features.conf will not touch parking lots that have calls still parked in them. Reload again at a later time. Misc additional fixes: * Added unit test for parking lot dialplan usage checking. * Made update connected line when a parked call is retrieved from a parking lot. * Made retrieved parked call stop ringing or MOH depending upon how the call was waiting in the parking lot. * Made CLI "features show" indicate if the parking lot is enabled for use. * Added PARKINGDYNEXTEN channel variable to allow dynamic parking lots to specify the parking lot access extension. * Made AMI ParkedCalls action ParkedCall events have a Parkinglot header. * Made AMI ParkedCalls action ParkedCallsComplete event have a Total header. * Fixed potential deadlock from AMI Park action holding channel locks while calling masq_park_call(). * Fixed several places where ast_strdupa() were used inside of loops. (Mostly fixed by refactoring the loop body into its own function.) * Fixed copy_parkinglot() copying too much from the source parking lot. Extracted the parking lot configuration settings into struct parkinglot_cfg. * Refactored courtesytone playing code to put the channel not playing the tone in autoservice. * Fix when pbx-parkingfailed is played that the other channel is put in autoservice if it exists. * Fixed parkinglot reference leak in parked_call_exec() error paths. * Fixed parkinglot_unref() use of parkinglot after it was unreffed. * Made destroy the struct ast_parkinglot parkings lock when done. * Refactored the features.conf parking lot configuration code to eliminate redundancy. * Fixed feature reload to better protect parking lots. * Fixed parking lot container reference leak in handle_parkedcalls(). * Fixed the total count in handle_parkedcalls(). Review: https://reviewboard.asterisk.org/r/1358/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-16Merged revisions 332029 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug 2011) | 2 lines Moved notes about 'storesipcause' to UPGRADE.txt from CHANGES AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-16Merged revisions 332022 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332022 | mnicholson | 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) | 16 lines In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) | 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-10Merged revisions 331418 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011) | 6 lines Revert -r318141. It was a band-aid that only partially fixed parking. A better fix is on reviewboard review 1358. (issue ASTERISK-17374) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-10SIP display-name needed to be empty for Avaya IP500Jonathan Rose
In order to address a compatability issue with certain features on certain devices which rely on display name content to change behavior, initreqprep in chan_sip.c has been changed to no longer substitute cid_number into the display name when cid_name isn't present. Instead, it will send no display name in that case. (closes issue ASTERISK-16198) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-08Merged revisions 331097 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) | 5 lines Bump the AMI protocol version to 1.2 As a result of converting Unlink events that were missed in the AMI 1.1 update to Bridge events, the AMI protocol version is being incremented. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-08Merged revisions 331041 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines Replace AMI Unlink events with Bridge events A previous update converted some of the Link and Unlink events to Bridge events, but a couple of Unlink events were missed. This patch rectifies the situation. (closes issues ASTERISK-17455) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-27reverting 329840 due to failing tests. Going to change this feature to be ↵Jonathan Rose
purely optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-27Adds cdr logging of calls resulting in CONGESTIONJonathan Rose
Applies a patch made a long time ago by alecdavis which adds a CDR feature for logging calls that failed due to congestion. (closes issue #15907) Reported by: alecdavis Patches: cdr_congestion.diff.txt uploaded by alecdavis (license #5546) Review: https://reviewboard.asterisk.org/r/454/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21Merged revisions 329257 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines s/1.10/10.0/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-15Merged revisions 328448 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) | 2 lines Update UPGRADE.txt and CHANGES files. Update documentation files stating that deprecated modules are no longer built by default. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-08Adds entry in UPDATES.txt for removal of formats/format_sln16.c. Fixes typo ↵David Vossel
in CHANGES as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-08Updates CHANGES log to reflect new slinear read/write file interpreters.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-07Fixes spelling errors in CHANGES as well as adding a few entries for CELT ↵David Vossel
and confbridge. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-06Replace Berkeley DB with SQLite 3Terry Wilson
There were some bugs in the very ancient version of Berkeley DB that Asterisk used. Instead of spending the time tracking down the bugs in the Berkeley code we move to the much better documented SQLite 3. Conversion of the old astdb happens at runtime by running the included astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave identically to the old Berkeley backend, but in the future we could offer a much more robust interface. We do not include the SQLite 3 library in the source tree, but instead rely upon the distribution-provided libraries. SQLite is so ubiquitous that this should not place undue burden on administrators. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-05New feature: AMI Action FilterAddMark Murawki
This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session (closes issue ASTERISK-16795) Reported by: kobaz Tested by: kobaz,loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-01Change CHANGES move the commits to the right placeGregory Nietsky
r296249 r318141 Application changes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-01Change CHANGES move the commits to the right place in the file missed in reviewGregory Nietsky
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-30Video support for ConfBridge.David Vossel
Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-30Fax gateway functionality (i.e. translating between a T.30 terminal and a T.38Matthew Nicholson
terminal). Can be enabled on a channel by setting FAXOPT(gateway)=yes in the dialplan. Big thanks to irroot for porting this code to use the framehooks api. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-29Commit "distrotech" app_queue changes to TrunkGregory Nietsky
* Added general option negative_penalty_invalid default off. when set members are seen as invalid/logged out when there penalty is negative. for realtime members when set remove from queue will set penalty to -1. * Added queue option autopausedelay when autopause is enabled it will be delayed for this number of seconds since last successful call if there was no prior call the agent will be autopaused immediately. * Added member option ignorebusy this when set and ringinuse is not will allow per member control of multiple calls as ringinuse does for the Queue. - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty. (closes issue ASTERISK-17421) (closes issue ASTERISK-17391) Reported by: irroot Tested by: irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15CONFBRIDGE_INFO function to get conference dataKinsey Moore
Added the CONFBRIDGE_INFO dialplan function to get information about a conference bridge including locked status and number of parties, admins, and marked users. Review: https://reviewboard.asterisk.org/r/1271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-13Addition of "outofcall_message_context" sip.conf option.David Vossel
Review: https://reviewboard.asterisk.org/r/1265/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-01Support routing text messages outside of a call.Russell Bryant
Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-27Merged revisions 321337 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-27Merged revisions 321330 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) | 8 lines The app_privacy args have undocumented "options" position, interferes with "context" position. * Add documention for unused "options" position to match existing code. The trunk(v1.10) version will remove the unused options position. (closes issue #19273) Reported by: mdavenport ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@321331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25Merged revisions 320823 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-25 CHANNEL(pickupgroup)Gregory Nietsky
Allow Setting / Reading the pickupgroup of a channel with func_channel.c (closes issue #19045) Reported by: irroot Review: https://reviewboard.asterisk.org/r/1148/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-23Merged revisions 320650 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 May 2011) | 16 lines Add ConnectedLineNum/Name headers to output of AMI action Status. * Add ConnectedLineNum and ConnectedLineName headers to the output of the AMI action Status. This makes it easier to find out who the channel is connected to without having to lookup BridgedChannel or when they are connected to an application (e.g.: VoiceMail) which has no bridged channel. * Bridged channels with no CallerID had "" instead of "<unknown>" output, that might be a bug as "<unknown>" was what older versions used. (closes issue #18158) Reported by: gareth Patches: svn-292308.diff uploaded by gareth (license 208) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-20Adds STRREPLACE functionJonathan Rose
Adds a new STRREPLACe function to func_strings.c that allows users to search and replace against a variable in the dialplan. (closes issue #18023) Reported by: wdoekes Review: https://reviewboard.asterisk.org/r/1219/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-16 When a error in T.38 negotiation happens or its rejected on a channel theGregory Nietsky
state of the channel reverts to unknown this should be rejected. this is important for negotiating T.38 gateway see #13405 This patch adds a option T38_REJECTED that behaves as T38_DISABLED except it reports state rejected. Trivial Change to res_fax to honnor UNAVAILABLE and REJECTED states. (closes issue #18889) Reported by: irroot Tested by: irroot, darkbasic, mnicholson Review: https://reviewboard.asterisk.org/r/1115 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@319087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-09Allows ParkedCall application to specify a parkinglot.Jonathan Rose
When invoking the app parkedcall, the argument can now include '@parkinglot' after the extension. (closes issue #18777) Reported by: cartama Patches: 0018777.diff uploaded by cartama (license 1157) Review: https://reviewboard.asterisk.org/r/1209/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-06Add the Uniqueid header to Userevent.Russell Bryant
(closes issue #16962) Reported by: jlpedrosa Patches: patch.diff uploaded by jlpedrosa (license 1002) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-06Updated CHANGES to note the autoservice changes for pbx_luaMatthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-06Use two spaces after periods for the recent pbx_lua change descriptionsMatthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-06Updated CHANGES for hints support in pbx_luaMatthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-06Detect Goto in pbx_lua.Matthew Nicholson
This code will actually detect any dialplan jump from any application that calls ast_explicit_goto(). This change is only being done in trunk as it may change the way some dialplans execute. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317721 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-05Add "calendar show types" CLI command.Russell Bryant
(closes issue #18246) Reported by: junky Patches: calendar_types.diff uploaded by junky (license 177) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-05Add CEL extra field to cel_pgsql.Russell Bryant
(closes issue #18462) Reported by: joscas Patches: bug_18462.diff uploaded by snuffy (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-04Reverts rev 316218 as it breaks parsing the [general] section of sip.conf.David Vossel
The functionality this patch attempts to achieve should already be possible using [general](+) in the config file. issue #17957 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-05-03If multiple [general] contexts occur from sip.conf (usually due to external ↵Tilghman Lesher
includes), merge them. The original implementation of this did the merging of all contexts with the same name in the realtime layer, but that implementation severely breaks drivers which use the same context name (e.g. iax.conf, type={peer,user}). Therefore, the implementation needs to do the merging for particular entries only, based upon what contexts would allow that in the channel driver itself. This implementation is for chan_sip only, but others could be added in the future. (closes issue #17957) Reported by: marcelloceschia Patches: chan-sip_parsing-general_branch162.patch uploaded by marcelloceschia (license 1079) Tested by: tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-21New HD ConfBridge conferencing application.David Vossel
Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-20Introduction of the JITTERBUFFER dialplan function.David Vossel
Review: https://reviewboard.asterisk.org/r/1157/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13Add 'description' field for CLI and Manager outputLeif Madsen
(closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-04Makes 'dialplan add extension' create the specified context if it does not ↵Jonathan Rose
already exist. If the user invokes 'dialplan add extension' into a non-existing context, the context will be created and a message informing the user of the context being created will be issued in cli. (closes issue #17431) Reported by: leearcher Patches: context_auto_create.diff uploaded by kobaz (license 834) Tested by: leearcher, kobaz, jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-01New Feature for chan_dahdi. 4 length pattern matching.Jonathan Rose
In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns. The s ntax remains the same and the method used to track the pattern history will only change when using the length 4 patterns. (closes issue SWP-3250) Code: jrose rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-18Adds an option to FollowMe that isn't useful for the bug it was made to ↵Jonathan Rose
solve. Still, due to the nature of FollowMe, it makes sense to have this option since it keeps apps bound to channels that would otherwise go away from being lost. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@311427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-11Mix Monitor: Now with r and t options.Jonathan Rose
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-04Add setvar option to calendaringTerry Wilson
Adding the setvar option with variable substitution on the value allows things like setting the outbound caller id name to the summary of a calendar event, etc. Values could be chained together as they are appended in order to do some scripting if necessary. Review: https://reviewboard.asterisk.org/r/1134/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-19 20:31:46 +0000