summaryrefslogtreecommitdiff
path: root/configs
AgeCommit message (Collapse)Author
2008-08-04Merged revisions 135536 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) | 2 lines fix a config sample typo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-04Merged revisions 135473 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) | 2 lines Add a minor clarification to the documentation of mohinterpret and mohsuggest ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-01Merge changes from team/bbryant/keyrotationRussell Bryant
This set of changes enhances IAX2 encryption support by adding key rotation to provide enhanced security. The key used for encryption is rotated right after the call gets set up, and then again every few minutes. This was discussed at the last AstriDevCon. For interoperability with older versions of Asterisk, there is an option that disables key rotation. (closes issue #13018) Reported by: bbryant Patches: 07072008__iax2_key_rotation.diff uploaded by bbryant (license 36) Tested by: russell, bbryant git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-01SIP should use the transport type set in the Moved Temporarily for the nextTilghman Lesher
invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-01IMAP storage functioned under the assumption that foldersMark Michelson
such as "Work" and "Family" would be subfolders of the INBOX. This is an invalid assumption to make, but it could be desirable to set up folders in this manner, so a new option for voicemail.conf, "imapparentfolder" has been added to allow for this. (closes issue #13142) Reported by: jaroth Patches: parentfolder.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-30Move implementation of an attended-transfer-complete sound from one channelTilghman Lesher
driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-28remove remaining Zaptel references in various placesKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-22Merged revisions 132713 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-22Merged revisions 132641 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) | 2 lines use renamed libpri API call for controlling this feature (was improperly named before) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-21Update configuration files to add missing options for jingle, gtalk, Brett Bryant
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
2008-07-15Additional option for videosupport (always) that disables the optimization toTilghman Lesher
fail to setup video RTP if the two endpoints will not support it. This assists with call files and certain transfers to ensure that if two video phones are ever connected, they will always share a video feed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11Merged revisions 130039 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today (related to issue #13042) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-07Update a few instances of "extensions reload" to "dialplan reload"Mark Michelson
in the documentation. Patch provided by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128599 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" ↵Olle Johansson
and "tlsbindaddr". Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06- Fixing issues with "sip show settings"Olle Johansson
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Make TCP disabled by default (it's considered experimental)Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Reformatting the config sampleOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Add option to wait to be able to explicitly send ACM via the Proceeding() ↵Matthew Fredrickson
application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-03Added a new option, "timeoutpriority" to queues.conf. A detailedMark Michelson
explanation of the change may be found in configs/queues.conf.sample (closes issue #12690) Reported by: atis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-02The ackcall and endcall options in agents.conf now have supplemental optionsMark Michelson
acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable instead of being hardcoded to '#' and '*'. (AST-86) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127558 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-01Add a configuration option so the global outboundproxy can use tcptls ↵Brett Bryant
without it being defined by each sip user. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-01Merged revisions 126844 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines Clear up documentation on "domain=" setting in sip.conf Reported by: davidw (closes issue #12413) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-30rename zapata.conf.sample to chan_dahdi.conf.sampleJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-27Change the way that the transport option works for sip users. transport will ↵Brett Bryant
now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-26Merged revisions 125218 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) | 4 lines Document ackcall=always. (closes issue #12852) Reported by: davidw ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-26Update sample configuration to match what are now the defaults for the prefix.Tilghman Lesher
(closes issue #12838, related to issue #12198) Reported by: pabelanger Patches: http.conf.diff2 uploaded by pabelanger (license 224) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@125191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-22Revert my change to the sample meetme conf file as it was incorrect.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-22Fix a comment in meetme.conf.sample per jmls via #asterisk-devSean Bright
(And this time, do it in the correct repository :-)) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-19Allow alternative extensions to be specified for a user.Tilghman Lesher
(closes issue #12830) Reported by: jcollie Patches: astertisk-trunk-121496-alternate-extensions.patch uploaded by jcollie (license 412) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-19Merged revisions 123883 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines Correct description of notifyringing option. (Closes issue #12890) Reported by gminet ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-16Note that only one timing interface should get loaded.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-12Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. ↵Jeff Peeler
Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-10Merge another big set of changes from team/russell/eventsRussell Bryant
This commit merges in the rest of the code needed to support distributed device state. There are two main parts to this commit. Core changes: - The device state handling in the core has been updated to understand device state across a cluster of Asterisk servers. Every time the state of a device changes, it looks at all of the device states on each node, and determines the aggregate device state. That resulting device state is what is provided to modules in Asterisk that take actions based on the state of a device. New module, res_ais: - A module has been written to facilitate the communication of events between nodes in a cluster of Asterisk servers. This module uses the SAForum AIS (Service Availability Forum Application Interface Specification) CLM and EVT services (Cluster Management and Event) to handle this task. This module currently supports sharing Voicemail MWI (Message Waiting Indication) and device state events between servers. It has been tested with openais, though other implementations of the spec do exist. For more information on testing distributed device state, see the following doc: - doc/distributed_devstate.txt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-10Update dundi.conf to indicate that the asterisk.conf entityid option can be usedRussell Bryant
to set the entityid used in DUNDi, as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-05Merge the adaptive realtime branch, which will make adding new required fieldsTilghman Lesher
to realtime less painful in the future. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-03Move compatibility options into asterisk.conf, default them to on for upgrades,Tilghman Lesher
and off for new installations. This includes the translation from pipes to commas for pbx_realtime and the EXEC command for AGI, as well as the change to the Set application not to support multiple variables at once. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-28Merged revisions 118646 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-27Merged revisions 118358 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008) | 3 lines Add a note that pbx_config.so is needed for Local channels. (Closes issue #12671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22Add a compatibility option for upgrading realtime extensionsTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22Minor text fix. roster -> resource.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-19Change the default for the pridialplan parameter to the far more common case ofTilghman Lesher
'unknown', and better document the use of each parameter. (closes issue #12633) Reported by: tzafrir Patches: pridialplan_unknown_2.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-19fix example configuration for video support in chan_ossLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Merged revisions 116409 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | 1 line Document exitcontext in app_voicemail sample config ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116410 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-10fix a sample since we now required , and not | for the arguments separatorClaude Patry
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115595 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-09Allow a password change to be validated by an external script.Tilghman Lesher
(closes issue #12090) Reported by: jaroth Patches: vm-check-newpassword.diff.txt uploaded by mvanbaak (license 7) 20080509__bug12090.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30Add support for specifying the registration expiry on a per registration ↵Joshua Colp
basis in the register line. This comes from a Switchvox patch. (issue AST-24) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30Adding new configuration options to app_queue. This adds two new valuesMark Michelson
to announce-position, "limit" and "more," as well as a new option, announce-position-limit. For more information on the use of these options, see CHANGES or configs/queues.conf.sample. (closes issue #10991) Reported by: slavon Patches: app_q.diff uploaded by slavon (license 288) Tested by: slavon, putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Add support for authenticating on a NOTIFY request. This is useful for ↵Joshua Colp
phones that require it when sending them a special packet to get them to do something (such as reload their configuration). (closes issue #9896) Reported by: IgorG Patches: sipnotify-113980-v14.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21(closes issue #6113)Jeff Peeler
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-16This is the scariest commit I've done in a long time. This is the ↵Steve Murphy
astobj2-ification of chan_sip. I've tested a number of scenarios like crazy. It used to have 4x the call setup/teardown performance of trunk, but now it's roughly at parity. I will attempt to find the bottlenecks and get it back to the 4x mark. The changes made were somewhat invasive, but the value to the community of these upgrades outweighs waiting further for more testing. Every change being made to chan_sip was lousing this code up when we tried to merge. Peers, Users, Dialogs, are all now astobj2 objects, indexed via hashtables. Refcounting is used to track objects and free them at the bitter end of their lives. Please file issues on bugs.digium.com, and PLEASE, please, please be patient. One natural advantage to all the hash-table work is that loading large sip.conf files full of thousands of peers now goes much faster. One more please: PLEASE help thrash this code and test it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-03 16:04:04 +0000