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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
2010-05-03Add new admin features to meetme: Roll call, eject all, mute all, record ↵jpeeler1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@260757 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Asterisk data retrieval API.eliel1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@258517 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21IAXpeers output now matches SIPpeers format for manager (AMI).lmadsen1-0/+2
(closes issue #17100) Reported by: secesh Tested by: pabelanger Review: https://reviewboard.asterisk.org/r/594/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-21Added CHANGES entry for new MixMonitorMute AMI command.jmls1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258227 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09func_srv and explicit specification of a remote IP for SIP.mmichelson1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@256485 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-29This patch adds custom device state handling for ConfBridge conferences,jsmith1-0/+2
matching the devstate handling of the MeetMe conferences. Review: https://reviewboard.asterisk.org/r/572/ Closes issue #16972 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@255281 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-24Allow configuration of minsecs and nextaftercmd per mailbox.jpeeler1-0/+1
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: http://svn.digium.com/svn/asterisk/trunk@254321 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Change per-file debug and verbose levels to be per-module, the waykpfleming1-0/+11
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: http://svn.digium.com/svn/asterisk/trunk@253917 f38db490-d61c-443f-a65b-d21fe96a405b
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