aboutsummaryrefslogtreecommitdiffstats
path: root/CHANGES
AgeCommit message (Collapse)AuthorFilesLines
2010-10-11Update CHANGES to reflect new gtalk.conf options.dvossel1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@291194 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-23Add note about the checkhangup option of ${CHANNEL()}tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@288606 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-20Addition of the FrameHook API (AKA AwesomeHooks)dvossel1-0/+1
So far all our tools for viewing and manipulating media streams within Asterisk have been entirely focused on audio. That made sense then, but is not scalable now. The FrameHook API lets us tap into and manipulate _ANY_ type of media or signaling passed on a channel present today or in the future. This tool is a step in the direction of expanding Asterisk's boundaries and will help generate some rather interesting applications in the future. In addition to the FrameHook API, a simple dialplan function exercising the api has been included as well. This function is called FRAME_TRACE(). FRAME_TRACE() allows for the internal ast_frames read and written to a channel to be output. Filters can be placed on this function to debug only certain types of frames. This function could be thought of as an internal way of doing ast_frame packet captures. Review: https://reviewboard.asterisk.org/r/925/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@287647 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-15Add parking extension for non-default parking lots.jpeeler1-0/+3
This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@286931 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-10Added missing documentation for ExternalIVR feature added in January 2010diruggles1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285992 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-09Merged revisions 285710 via svnmerge from bbryant1-4/+13
https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-03Sneak FIELDNUM() into 1.8. Returns a 1-based index into a list of a ↵tilghman1-0/+1
specified item. Matches up with FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth Patches: svn-279754.diff uploaded by gareth (license 208) Tested by: gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280809 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Updated documentation for FAX logger level.pabelanger1-2/+9
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279689 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-26Add documentation for FAX logger level.pabelanger1-0/+2
(closes issue #17715) Reported by: vrban Patches: 17715.patch uploaded by pabelanger (license 224) Tested by: vrban git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@279566 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merge the realtime failover branchtilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278957 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Separate queue_log arguments into separate fields, and allow the text file ↵tilghman1-0/+5
to be used, even when realtime is used. (closes issue #17082) Reported by: coolmig Patches: 20100720__issue17082.diff.txt uploaded by tilghman (license 14) Tested by: coolmig git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Add ability to configure the Max-Forwards header in the dialplan, as well as inoej1-0/+2
sip.conf configuration for the channel and for devices. The Max-Forwards header is used to prevent loops in a SIP network. Each intermediary, like SIP proxys and SBCs, decrement this counter and detects when it reaches zero, at which point the SIP request is nicely killed in a SIP-friendly way. Review: https://reviewboard.asterisk.org/r/778/ Thanks to dvossel for the review and good advice. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276951 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Add a dialplan function to check if a queue exists: QUEUE_EXISTSoej1-0/+1
Review: https://reviewboard.asterisk.org/r/777/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276950 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-13FILE() now supports line-mode and writing (altering) files.tilghman1-0/+1
(closes issue #16461) Reported by: skyman Patches: 20100622__issue16461.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/737/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276114 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-10Make indentation consistent, move some queue features to the queue section.russell1-54/+54
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275467 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-10Add support for devices with less than 3 lines on the LCD.russell1-1/+2
(closes issue #17600) Reported by: minaguib Patches: ast_unistim_height_v2.patch uploaded by minaguib (license 1078) Tested by: minaguib git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275466 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-09Include rdnis in msgXXXX.txt file.pabelanger1-0/+1
(closes issue #17566) Reported by: outcast Patches: voicemail-rdnis.patch uploaded by outcast (license 1071) Tested by: outcast git-svn-id: http://svn.digium.com/svn/asterisk/trunk@275307 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Add IPv6 to Asterisk.mmichelson1-0/+2
This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274783 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-07Also run the externnotify script when the pollmailboxes thread notices a change.tilghman1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274491 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Add regular expression filtering for manager events.jpeeler1-0/+2
This patch as documented in the sample config allows one to optionally apply white, black, or both types of filtering to manager events. The new 'eventfilter' option is set per user. (closes issue #14861) Reported by: fnordian Patches: eventfilter3.patch uploaded by fnordian (license 110), modified by me Review: https://reviewboard.asterisk.org/r/673/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271868 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Updated the CHANGES file documenting the addition of a configurable port in ↵mnicholson1-0/+1
the dundi config file. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271764 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-21Add new application for declining counting words in multiple languages.tilghman1-0/+3
(closes issue #16869) Reported by: chappell Patches: app_say_counted-20100317.c uploaded by chappell (license 8) Tested by: chappell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271520 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17adds support for slin16 in sipdvossel1-0/+1
(closes issue #16153) Reported by: kfister Patches: 16153-1.6.2.0-rc5.patch uploaded by kfister (license 912) slin16.sip.patch.1 uploaded by malcolmd (license 924) Tested by: kfister, malcolmd git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271261 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-17adds speex 16khz audio supportdvossel1-0/+2
(closes issue #17501) Reported by: fabled Patches: asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448) Tested by: malcolmd, fabled, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271231 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16addition of G.719 pass-through supportdvossel1-0/+3
(closes issue #16293) Reported by: malcolmd Patches: g719.passthrough.patch.7 uploaded by malcolmd (license 924) format_g719.c uploaded by malcolmd (license 924) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270940 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16MSG_OOB flag on HANGUP packet removed.pabelanger1-1/+2
Per Tilghman's request on IRC (#asterisk-bugs). (closes issue #17506) Reported by: brycebaril Tested by: pabelanger, tilghman git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270936 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-15Add distributed devicestate via the XMPP protocol.tilghman1-0/+2
(closes issue #15757) Reported by: Marquis Patches: distributed_devstate-XMPP.txt uploaded by lmadsen (license 10) Tested by: Marquis, lmadsen, marcelloceschia Review: https://reviewboard.asterisk.org/r/351/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270519 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-11Add DBGetComplete event after a DBGetResponse.tilghman1-0/+2
(closes issue #16965) Reported by: rrb3942 Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269938 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-09dial by name in chan_dahditzafrir1-0/+7
* chan_dahdi supports dialing configuring and dialing by device file name. DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise it may appear in chan_dahdi.conf as 'channel => span-name!local!1'. * A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean. False by default. If set, chan_dahdi will ignore failed 'channel' entries. Handy for the above name-based syntax as it does not depend on initialization order. * have my_pri_make_cc_dialstring() only manupulate dial-strings of group (gGrR) dialing, which make it lsightly more complicated. https://reviewboard.asterisk.org/r/535/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269238 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Add High Resolution Times to CDRs for Asterisksnuffy1-0/+5
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change. Patch by snuffy. (closes issue #16559) Reported by: cianmaher Tested by: cianmaher, snuffy Review: https://reviewboard.asterisk.org/r/461/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@269153 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-08Add SRTP support for Asterisktwilson1-0/+13
After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268894 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-05Typo fix.kpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268417 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-05Grammatical error fix.kpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268395 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Update UPGRADE.txt and CHANGE for CDR functionality changes.lmadsen1-0/+6
Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity written unless cdr.conf exists and is configured. (closes issue #17373) Reported by: wdoekes Tested by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267624 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-03Add ETSI Message Waiting Indication (MWI) support.rmudgett1-0/+1
Add the ability to report waiting messages to ISDN endpoints (phones). Relevant specification: EN 300 650 and EN 300 745 Review: https://reviewboard.asterisk.org/r/599/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267399 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Malicious Call ID support.rmudgett1-0/+1
Add the ability to report malicious callers as an AMI event in the call event class. Relevant specification: EN 300 180 Review: https://reviewboard.asterisk.org/r/576/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267350 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Call Waiting support.rmudgett1-0/+7
Add the ability to announce a call to an endpoint when there are no B channels available. A call waiting call is a SETUP message with no B channel selected. Relevant specification: EN 300 056, EN 300 057, EN 300 058 For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the "no_media_path" option. * Returns "0" if there is a B channel associated with the call. * Returns "1" if no B channel is associated with the call. The call is either on hold or is a call waiting call. If you are going to allow incoming call waiting calls then you need to use CHANNEL(no_media_path) do determine if you must drop a call to accept the new call. Review: https://reviewboard.asterisk.org/r/568/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267261 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Update CHANGES and aoc help doc to reflect AOC additionsdvossel1-0/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267181 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Advice Of Charge (AOC) event reporting.rmudgett1-0/+2
This feature generates AMI events in the new aoc event class from the events passed up by libpri. Review: https://reviewboard.asterisk.org/r/537/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267008 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Explicit Call Transfer (ECT) support.rmudgett1-0/+4
Added ability to send and receive ETSI Explicit Call Transfer (ECT) messages to eliminate tromboned calls. Note: Asterisk already supported initiating the transfer of calls to eliminate tromboned calls to libpri so there was nothing to do for the asterisk portion. Review: https://reviewboard.asterisk.org/r/520/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266926 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-01Support setting locale per-mailbox (changes date/time languages for email, ↵tilghman1-0/+1
pager messages). (closes issue #14333) Reported by: klaus3000 Patches: 20090515__issue14333.diff.txt uploaded by tilghman (license 14) app_voicemail.c-svn-trunk-rev211675-patch.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266828 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-27Cache query results for one second.tilghman1-0/+2
Queries from the PBX core come in 3's. Caching avoids the additional performance penalty from those two additional queries hitting the database. (closes issue #16521) Reported by: tilghman Patches: 20091229__issue16521.diff.txt uploaded by tilghman (license 14) Tested by: Hubguru, tilghman git-svn-id: http://svn.digium.com/svn/asterisk/trunk@266238 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-24Calendaring support for Exchange Server 2007+ via EWStwilson1-2/+4
This commit adds support for calendaring with Exchange Server 2007+ via Exchange Web Services. Full write support and for querying attendees. Many thanks to Jan Kaláb for the feature. (closes issue #17022) Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel (license 1008) Tested by: pitel, twilson Review: https://reviewboard.asterisk.org/r/557/ Review: https://reviewboard.asterisk.org/r/668/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265317 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Add support for direct media ACLstwilson1-0/+2
directmediapermit/directmediadeny support to restrict which peers can do directmedia based on ip address. In some networks not all phones are fully routed, i.e. not all phones can ping each other. This patch adds a way to restrict directmedia for certain peers between certain networks. (closes issue #16645) Reported by: raarts Patches: directmediapermit.patch uploaded by raarts (license 937) Tested by: raarts Review: https://reviewboard.asterisk.org/r/467/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264626 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-19Add ability to hangup all channels from the CLI.lmadsen1-0/+2
Added the keyword 'all' to the 'channel hangup request' CLI command so that you can request all channels to be hungup without having to restart Asterisk. (closes issue #16009) Reported by: moy Patches: hangup-all-rev-221688.patch uploaded by moy (license 222) Tested by: moy, russell git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264117 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-18put changes with the correct versionjpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263808 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-18Merged revisions 263769 via svnmerge from jpeeler1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines Modify directory name reading to be interrupted with operator or pound escape. In the case of accidentally entering the wrong first three letters for the reading, users could be very frustrated if the name listing is very long. This allows interrupting the reading by pressing 0 or #. 0 will attempt to execute a configured operator (o) extension and # will exit and proceed in the dialplan. ABE-2200 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263807 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Update CHANGES to reflect DAHDI buffer dialstring option backport to 1.6.2dvossel1-9/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@263294 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05New 'manager show settings' CLI command.pabelanger1-0/+2
See the CHANGES file for more details. (closes issue #16343) Reported by: pabelanger Patches: issue16343.patch.v5 uploaded by pabelanger (license 224) Tested by: pabelanger, tilghman, lmadsen Review: https://reviewboard.asterisk.org/r/630/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261180 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Add new possible value to autopause option to allow members to be autopaused ↵mmichelson1-0/+4
in all queues. See the CHANGES file and queues.conf.sample for more details. (closes issue #17008) Reported by: jlpedrosa Patches: queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002) Review: https://reviewboard.asterisk.org/r/581/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261051 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-08 09:50:46 +0000