asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	58f281a670	Add "setvar" option to manager.conf. With this option set, channel variables can be set on every manager originate. The Variable header can still be used to set additional channel variables for individual calls if desired. This work was completed by Olle Johansson on review board. I have applied the review feedback and am committing it in order to get this into trunk before Asterisk 11 is branched. Review: https://reviewboard.asterisk.org/r/1412 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370681 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	10afdf3a2a	Named ACLs: Introduces a system for creating and sharing ACLs This patch adds Named ACL functionality to Asterisk. This allows system administrators to define an ACL and refer to it by a unique name. Configurable items can then refer to that name when specifying access control lists. It also includes updates to all core supported consumers of ACLs. That includes manager, chan_sip, and chan_iax2. This feature is based on the deluxepine-trunk by Olle E. Johansson and provides a subset of the Named ACL functionality implemented in that branch. For more information on this feature, see acl.conf and/or the Asterisk wiki. Review: https://reviewboard.asterisk.org/r/1978/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	14a985560e	Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	0f4489dc0f	Fix TLS port binding behavior as well as reload behavior: * Removes references to tlsbindport from http.conf.sample and manager.conf.sample * Properly bind to port specified in tlsbindaddr, using the default port if specified. * On a reload, properly close socket if the service has been disabled. A note has been added to UPGRADE.txt to indicate how ports must be set for TLS. (closes issue ASTERISK-16959) reported by Olaf Holthausen (closes issue ASTERISK-19201) reported by Chris Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas Review: https://reviewboard.asterisk.org/r/1709 ........ Merged revisions 353770 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 353820 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353821 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	3b53a9cdb3	Merged revisions 332817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 \| mjordan \| 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) \| 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	a77fd545ab	Merged revisions 312766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r312766 \| mnicholson \| 2011-04-05 09:14:50 -0500 (Tue, 05 Apr 2011) \| 22 lines Merged revisions 312764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r312764 \| mnicholson \| 2011-04-05 09:13:07 -0500 (Tue, 05 Apr 2011) \| 15 lines Merged revisions 312761 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r312761 \| mnicholson \| 2011-04-05 09:10:34 -0500 (Tue, 05 Apr 2011) \| 8 lines Limit the number of unauthenticated manager sessions and also limit the time they have to authenticate. AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested by: mnicholson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	5673d0855e	Merged revisions 291230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291230 \| lmadsen \| 2010-10-12 11:08:04 -0500 (Tue, 12 Oct 2010) \| 10 lines Merged revisions 291229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r291229 \| lmadsen \| 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010) \| 2 lines Add documention that mentions options are defined but not used. (Issue #18101) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291231 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c48c7123c1	Merged revisions 283627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283627 \| russell \| 2010-08-26 07:26:22 -0500 (Thu, 26 Aug 2010) \| 2 lines Move httptimeout out from in between port and bindaddr. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
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
Richard Mudgett	afd4454c44	Generic Advice of Charge. Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267096 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
Matthew Nicholson	2724f89bba	Merged revisions 257070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257070 \| mnicholson \| 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) \| 9 lines Add an option to restore past broken behavor of the Events manager action Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested. (closes issue #17023) Reported by: nblasgen Review: https://reviewboard.asterisk.org/r/602/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257146 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
Tilghman Lesher	5e2aa190fe	Display a list of channel variables in each channel-oriented event. (Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	f22962a0c1	Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197535 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	a7d813cae7	Remove a bunch of trailing whitespace in preparation for reformatting/cleanup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ca138fc807	Consistent SSL/TLS options across conf files ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though. Review: http://reviewboard.digium.com/r/237/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8f0b88c8c8	TLS/SSL private key option Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up. Review: http://reviewboard.digium.com/r/234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	ea6f754d4d	Update configuration files to add missing options for jingle, gtalk, manager.conf, and features.conf. (closes issue #13128) Reported by: caio1982 Patches: missing_options1.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	36cd3d0107	Additional note git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	763da3332a	Document 'originate' permission in manager sample config. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	857e3412f4	Several manager changes: 1) Add the Dialplan class, for NewExten and VarSet events, which should cut down on the volume of traffic in the Call class. 2) Permit some commands to be run from multiple classes, such as allowing DBGet to be run from either the System or the Reporting class. 3) Heavily document each class in the sample config, as there were several that made no sense to be in the write= line, and two that made no sense to be in the read= line (since they controlled no permissions there). (Closes issue #10386) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97651 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	ce2f670228	Change cdr_manager to use a "CDR" level, rather than the (overcrowded) "call" level. (Closes issue #11015) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Jason Parker	7a1c2d94bb	Add manager events for RTCP statistics. Also adds a new "reporting" permission for manager, since it can be incredibly spammy. This permission was discussed on the -dev mailing list some months back. Issue 8613, patch by johann8384, with some minor changes by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Jason Parker	bd3de6d0f1	Change displayconnects option in manager.conf to be per-user. Issue 9932, patch by eliel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	7ee1303796	Add an option that lets you only allow one connection at a time for each manager user. (issue #8664, reported and original patch by ssokol, patch updated by bkruse, and further updated by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64786 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Pari Nannapaneni	f1ca07ea8b	Merged revisions 63047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63047 \| pari \| 2007-05-04 11:45:29 -0500 (Fri, 04 May 2007) \| 1 line explanation for httptimeout in manager.conf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	47dd5a15af	Issue 6082 - New DTMF event for manager git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	437f4288cd	- Generalize the function ssl_setup() so that the certificate info are passed as an argument. - Update the code in main/http.c to use the new interface (the diff is large but mostly mechanical, due to the name change of several variables); - And since now it is trivial, implement "AMI over TLS", and document the possible options in manager.conf - And since the test client (openssl s_client -connect host:port ) does not generate \r\n as a line terminator, make get_input() also accept just a \n as a line terminator (Mac users: do you also need the \r-only version ?) The option parsing in manager.conf is not very efficient, and needs to be cleaned up and made similar to what we have in http.conf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	0b8669b87e	document the "debug" parameter, and the change manager list -> manager show git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	d171c3d864	remove unused fields and unimplemented options. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45518 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Steve Murphy	6e42aa676c	CLI reverbification update to this config file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Mark Spencer	837910062b	First pass at in-place file manipulation via manager git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Olle Johansson	5237a0e06d	- Use systemname for realm in sip, if we have no configuration for realm - Optionally send systemname in manager (cool when you have a manager proxy) - Use systemname in CLI prompt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Mark Spencer	bfba044b5f	Flesh out the remainder of the manager + http changes and create a sample application to partially demonstrate the capability of manager over http. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@16850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	a1a5e746ee	add optional timestamps to manager events (issue #5535 , simplified) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	2c65582b66	remove extraneous svn:executable properties git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	f387421905	make sample config files easier to ready (issue #5371 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Kevin P. Fleming	f7d567fe85	allow write timeout to be set on a per-user basis in AMI (issue #5352 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago
Mark Spencer	3fa3e8539d	Merge config sample updates mentioning reload (bug #3697 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	21 years ago
Mark Spencer	8ddb3aba81	Allow connection notifications on manager interface to be hidden (bug #3085 ) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	21 years ago
Mark Spencer	fda8b94bdf	Add access control to management interface git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	23 years ago
Mark Spencer	21dd479a6b	Fix sample config, too git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	23 years ago
Mark Spencer	105b71e710	Version 0.2.0 from FTP git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	23 years ago

44 Commits (a673c9ed1d2666a04617440385e481fc69458a1d)