aboutsummaryrefslogtreecommitdiffstats
path: root/configs/manager.conf.sample
AgeCommit message (Collapse)AuthorFilesLines
2010-10-12Merged revisions 291230 via svnmerge from lmadsen1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@291231 f38db490-d61c-443f-a65b-d21fe96a405b
2010-08-26Merged revisions 283627 via svnmerge from russell1-8/+10
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: http://svn.digium.com/svn/asterisk/trunk@283628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-22Add regular expression filtering for manager events.jpeeler1-0/+16
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
2010-06-02Generic Advice of Charge.rmudgett1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Add ETSI Advice Of Charge (AOC) event reporting.rmudgett1-0/+1
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
2010-04-13Merged revisions 257070 via svnmerge from mnicholson1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@257146 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09Merge Call completion support into trunk.mmichelson1-0/+3
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
2009-11-13Display a list of channel variables in each channel-oriented event.tilghman1-0/+6
(Closes AST-33) Reviewboard: https://reviewboard.asterisk.org/r/368/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.seanbright1-10/+10
Let's try that again, this time removing trailing whitespace and not leading whitespace. I can't believe no one noticed. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197535 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Remove a bunch of trailing whitespace in preparation for reformatting/cleanup.seanbright1-3/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197528 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-29Consistent SSL/TLS options across conf filesdvossel1-9/+8
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: http://svn.digium.com/svn/asterisk/trunk@191028 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24TLS/SSL private key optiondvossel1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@190545 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-21Update configuration files to add missing options for jingle, gtalk, bbryant1-0/+1
manager.conf, and features.conf. (closes issue #13128) Reported by: caio1982 Patches: missing_options1.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132514 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-07Additional notetilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@113245 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-07Document 'originate' permission in manager sample config.qwell1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@113243 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-10Several manager changes:tilghman1-2/+24
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: http://svn.digium.com/svn/asterisk/trunk@97651 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Change cdr_manager to use a "CDR" level, rather than the (overcrowded) ↵tilghman1-2/+2
"call" level. (Closes issue #11015) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91173 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-21Add manager events for RTCP statistics.qwell1-2/+2
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: http://svn.digium.com/svn/asterisk/trunk@70961 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-11Change displayconnects option in manager.conf to be per-user.qwell1-0/+2
Issue 9932, patch by eliel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@68831 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-17Add an option that lets you only allow one connection at a time for eachrussell1-0/+4
manager user. (issue #8664, reported and original patch by ssokol, patch updated by bkruse, and further updated by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64786 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-04Merged revisions 63047 via svnmerge from pari1-0/+7
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: http://svn.digium.com/svn/asterisk/trunk@63105 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-10Issue 6082 - New DTMF event for managertilghman1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61324 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-07- Generalize the function ssl_setup() so that the certificate inforizzo1-0/+12
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: http://svn.digium.com/svn/asterisk/trunk@48351 f38db490-d61c-443f-a65b-d21fe96a405b
2006-11-04document the "debug" parameter, and the changerizzo1-2/+4
manager list -> manager show git-svn-id: http://svn.digium.com/svn/asterisk/trunk@47184 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-18remove unused fields and unimplemented options.rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@45518 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-04CLI reverbification update to this config filemurf1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44352 f38db490-d61c-443f-a65b-d21fe96a405b
2006-07-19First pass at in-place file manipulation via managermarkster1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37936 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-11- Use systemname for realm in sip, if we have no configuration for realmoej1-0/+1
- Optionally send systemname in manager (cool when you have a manager proxy) - Use systemname in CLI prompt git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26884 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-01Flesh out the remainder of the manager + http changes and create a sample ↵markster1-1/+8
application to partially demonstrate the capability of manager over http. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@16850 f38db490-d61c-443f-a65b-d21fe96a405b
2006-01-03add optional timestamps to manager events (issue #5535, simplified)kpfleming1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7750 f38db490-d61c-443f-a65b-d21fe96a405b
2005-11-29git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7221 ↵kpfleming1-0/+0
f38db490-d61c-443f-a65b-d21fe96a405b
2005-10-04make sample config files easier to ready (issue #5371)kpfleming1-12/+8
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6720 f38db490-d61c-443f-a65b-d21fe96a405b
2005-10-04allow write timeout to be set on a per-user basis in AMI (issue #5352)kpfleming1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6716 f38db490-d61c-443f-a65b-d21fe96a405b
2005-03-17Merge config sample updates mentioning reload (bug #3697)markster1-1/+21
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@5187 f38db490-d61c-443f-a65b-d21fe96a405b
2005-01-05Allow connection notifications on manager interface to be hidden (bug #3085)markster1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@4672 f38db490-d61c-443f-a65b-d21fe96a405b
2003-05-05Add access control to management interfacemarkster1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@961 f38db490-d61c-443f-a65b-d21fe96a405b
2003-04-14Fix sample config, toomarkster1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@847 f38db490-d61c-443f-a65b-d21fe96a405b
2002-09-02Version 0.2.0 from FTPmarkster1-0/+12
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@512 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-24 13:31:26 +0000