asterisk

Commit Graph

Author	SHA1	Message	Date
Jeff Peeler	42c24b585a	Add regular expression filtering for manager events. 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: https://origsvn.digium.com/svn/asterisk/trunk@271868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	519b5a09e4	Updated the CHANGES file documenting the addition of a configurable port in the dundi config file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	63fd368411	Add new application for declining counting words in multiple languages. (closes issue #16869) Reported by: chappell Patches: app_say_counted-20100317.c uploaded by chappell (license 8) Tested by: chappell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@271520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	ba3d1ad680	adds support for slin16 in sip (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: https://origsvn.digium.com/svn/asterisk/trunk@271261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b00f58da25	adds speex 16khz audio support (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: https://origsvn.digium.com/svn/asterisk/trunk@271231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	fcb055fb4e	addition of G.719 pass-through support (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: https://origsvn.digium.com/svn/asterisk/trunk@270940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	0bf94685fd	MSG_OOB flag on HANGUP packet removed. Per Tilghman's request on IRC (#asterisk-bugs). (closes issue #17506) Reported by: brycebaril Tested by: pabelanger, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	81c15adfa2	Add distributed devicestate via the XMPP protocol. (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: https://origsvn.digium.com/svn/asterisk/trunk@270519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d66b4616f0	Add DBGetComplete event after a DBGetResponse. (closes issue #16965) Reported by: rrb3942 Patches: DBGetComplete.patch uploaded by rrb3942 (license 1003) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tzafrir Cohen	6d627b8c38	dial by name in chan_dahdi * 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: https://origsvn.digium.com/svn/asterisk/trunk@269238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Bradley Latus	4405813297	Add High Resolution Times to CDRs for Asterisk 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: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	857814f435	Add SRTP support for Asterisk 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: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ade79c6671	Typo fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	e853c0d978	Grammatical error fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	dfa82e0852	Update UPGRADE.txt and CHANGE for CDR functionality changes. 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: https://origsvn.digium.com/svn/asterisk/trunk@267624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	1c67f208a7	Add ETSI Message Waiting Indication (MWI) support. 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: https://origsvn.digium.com/svn/asterisk/trunk@267399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	0760f4e70a	Add ETSI Malicious Call ID support. 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: https://origsvn.digium.com/svn/asterisk/trunk@267350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	afcbc93dae	Add ETSI Call Waiting support. 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: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	3280a5c0af	Update CHANGES and aoc help doc to reflect AOC additions git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	28264c52b9	Add ETSI Advice Of Charge (AOC) event reporting. 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: https://origsvn.digium.com/svn/asterisk/trunk@267008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	48dd4d1249	Add ETSI Explicit Call Transfer (ECT) support. 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: https://origsvn.digium.com/svn/asterisk/trunk@266926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	b0357dcc3e	Support setting locale per-mailbox (changes date/time languages for email, 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: https://origsvn.digium.com/svn/asterisk/trunk@266828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	05a1ec8abc	Cache query results for one second. 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: https://origsvn.digium.com/svn/asterisk/trunk@266238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	880cde12ac	Calendaring support for Exchange Server 2007+ via EWS 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: https://origsvn.digium.com/svn/asterisk/trunk@265317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	c7303d840e	Add support for direct media ACLs 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: https://origsvn.digium.com/svn/asterisk/trunk@264626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	a8a1961be7	Add ability to hangup all channels from the CLI. 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: https://origsvn.digium.com/svn/asterisk/trunk@264117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	115f5076f5	put changes with the correct version git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	94df424e1d	Merged revisions 263769 via svnmerge from 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: https://origsvn.digium.com/svn/asterisk/trunk@263807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	96d3e573c9	Update CHANGES to reflect DAHDI buffer dialstring option backport to 1.6.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Paul Belanger	b2f59bea24	New 'manager show settings' CLI command. 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: https://origsvn.digium.com/svn/asterisk/trunk@261180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	fc652b869a	Add new possible value to autopause option to allow members to be autopaused 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: https://origsvn.digium.com/svn/asterisk/trunk@261051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	8ddd92f823	Add new admin features to meetme: Roll call, eject all, mute all, record in-conf This patch adds the following in-conference admin DTMF features: 81 - Roll call (or simply user count if INTROUSER isn't enabled) 82 - Eject all non-admins 83 - Mute/unmute all non-admins 84 - Start recording the conference on the fly FWIW, this code uses newly recorded prompts. (closes issue #16379) Reported by: rfinnie Patches: meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940) modified slightly by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Eliel C. Sardanons	a753e8878b	Asterisk data retrieval API. This module implements an abstraction for retrieving and exporting asterisk data. Developed by: Brett Bryant <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY) <eliels@gmail.com> For the Google Summer of code 2009 Project. Documentation can be found in doxygen format and inside the header include/asterisk/data.h Review: https://reviewboard.asterisk.org/r/275/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	bb2fa21ac1	IAXpeers output now matches SIPpeers format for manager (AMI). (closes issue #17100) Reported by: secesh Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/594/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Julian Lyndon-Smith	5f32984fcb	Added CHANGES entry for new MixMonitorMute AMI command. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Mark Michelson	6cad0f1602	func_srv and explicit specification of a remote IP for SIP. From Review Board: There are two interrelated changes here. First, there is the introduction of func_srv. This adds two new read-only dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV records instead. In order to facilitate this work, I added a couple of new API calls to srv.h. ast_srv_get_record_count tells the number of records returned by an SRV lookup. This number is calculated at the time of the SRV lookup. ast_srv_get_nth_record allows one to get a numbered SRV record. Second, there is the modification to chan_sip that allows one to specify a hostname or IP address (along with a port) to send an outgoing INVITE to when dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV records and then use the host and port from the results to dial via a specific host instead of what is configured in sip.conf. Review: https://reviewboard.asterisk.org/r/608 SWP-1200 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jared Smith	c34ec47577	This patch adds custom device state handling for ConfBridge conferences, matching the devstate handling of the MeetMe conferences. Review: https://reviewboard.asterisk.org/r/572/ Closes issue #16972 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	560d5c6099	Allow configuration of minsecs and nextaftercmd per mailbox. Previously only configurable globally. A unit test has also been written to provide protection against parse failures for supported mailbox options. (closes issue #16864) Reported by: kobaz Patches: voicemail2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/555/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Kevin P. Fleming	ae6008ef3a	Change per-file debug and verbose levels to be per-module, the way users expect them to work. 'core set debug' and 'core set verbose' can optionally change the level for a specific filename; however, this is actually for a specific source file name, not the module that source file is included in. With examples like chan_sip, chan_iax2, chan_misdn and others consisting of multiple source files, this will not lead to the behavior that users expect. If they want to set the debug level for chan_sip, they want it set for all of chan_sip, and not to have to also set it for reqresp_parser and other files that comprise the chan_sip module. This patch changes this functionality to be module-name based instead of file-name based. To make this work, some Makefile modifications were required to ensure that the AST_MODULE definition is present in each object file produced for each module as well. Review: https://reviewboard.asterisk.org/r/574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	f468595789	PITCH_SHIFT dialplan function The PITCH_SHIFT function can be used on a channel to independently modify the pitch of both rx and tx audio streams. Now you can improve your conference calls by assigning a random pitch effect to everyone entering a meetme room, or just make your day more interesting by making your co-workers sound funny. These are just some of the numerious practical uses for this function. Enjoy! https://reviewboard.asterisk.org/r/526/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	6bd57e0720	Add new config option to control AMI alarm event reporting in chan_dahdi. New config parameter "reportalarms" added in chan_dahdi.conf which supports the following possible values: "channels": report each channel alarms (current behavior, default for backward compatibility) "spans": report an "SpanAlarm" event when the span of any configured channel is alarmed "all": report channel and span alarms (aggregated behavior) "none": do not report any alarms (closes issue #16709) Reported by: nahuelgreco Patches: chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250392 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	8ef8706944	Updated CHANGES file to mention res_fax and res_fax_spandsp. Also fixed MODULEINFO depends and conflicts for app_fax, res_fax, and res_fax_spandsp. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250302 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	b5c98d640a	adds 'p' option to PickupChan The 'p' option allows the PickupChan app to pickup a ringing phone by looking for the first match to a partial channel name rather than requiring a full match. (closes issue #16613) Reported by: syspert Patches: pickipbycallid.patch uploaded by syspert (license 938) pickupbycallerid_v2.patch uploaded by dvossel (license 671) Tested by: dvossel, syspert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	bb3792a8a7	Add new application VMSayName for use with voicemail. VMSayName that will play the recorded name of the voicemail user if it exists, otherwise will play the mailbox number. A unit test has been written to verify correct functionality called test_voicemail_vmsayname. (closes issue #14973) Reported by: ghjm Review: https://reviewboard.asterisk.org/r/530/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249889 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Philippe Sultan	945529cae8	Add a new manager event for our buddies status. The new JabberStatus event gives a concise view of the status change to the AMI clients. Thanks fiddur! (closes issue #16760) Reported by: fiddur Patches: 244498.2.diff uploaded by fiddur (license 678) Tested by: fiddur, phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	27a4cda821	Add support for GROUP_MATCH_COUNT regex matching on category Current support for regex matching was previously only available on the group. Also, error reporting for regex failures has been added. In addition to this feature enhancement a unit test has been written to check the regular expression logic to ensure the count operation is working as expected. (closes issue #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/503/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	57c819fd5e	addition of dynamic parkinglots feature This feature allows for parkinglots to be created dynamically within the dialplan. Thanks to all who were involved with getting this patch written and tested! (closes issue #15135) Reported by: IgorG Patches: features.dynamic_park.v3.diff uploaded by IgorG (license 20) 2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7) dynamic_parkinglot.diff uploaded by dvossel (license 671) Tested by: eliel, IgorG, acunningham, mvanbaak, zktech Review: https://reviewboard.asterisk.org/r/352/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
TransNexus OSP Development	034a79c303	Updated doc for OSP lookup application. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jeff Peeler	b527525ffc	Add some additional option support for non-default parking lots. The options are: parkedcallparking, parkedcallhangup, parkedcallrecording, and parkedcalltransfers. Previously these options were only available for the default parking lot. (closes issue #16641) Reported by: bluecrow76 Patches: asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76 (license 270) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago

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565 Commits (3f00e3ff039f2bae66e8155ece55afbfd7380d5e)