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${ noResults }
1046 Commits (46c3a5373952f87f1c772667172cef510e6ff7f3)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Olle Johansson
|
1626397996 |
Merged revisions 126844 via svnmerge from
|
18 years ago |
Jeff Peeler
|
8f216ea83a |
rename zapata.conf.sample to chan_dahdi.conf.sample
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126675 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Brett Bryant
|
12d5cebea2 |
Change the way that the transport option works for sip users. transport will 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 |
18 years ago |
Tilghman Lesher
|
e903ae0e91 |
Merged revisions 125218 via svnmerge from
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 |
18 years ago |
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Tilghman Lesher
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4da51cf496 |
Update sample configuration to match what are now the defaults for the prefix.
(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 |
18 years ago |
Sean Bright
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d3aa30e803 |
Revert my change to the sample meetme conf file as it was incorrect.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124669 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Sean Bright
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f10caa9500 |
Fix a comment in meetme.conf.sample per jmls via #asterisk-dev
(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 |
18 years ago |
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Tilghman Lesher
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122486b263 |
Allow alternative extensions to be specified for a user.
(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 |
18 years ago |
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Tilghman Lesher
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48a9e5cada |
Merged revisions 123883 via svnmerge from
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 |
18 years ago |
Russell Bryant
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63bb6565d0 |
Note that only one timing interface should get loaded.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122977 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jeff Peeler
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ef3b214728 |
Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. 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 |
18 years ago |
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Russell Bryant
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e9d72e0cb2 |
Merge another big set of changes from team/russell/events
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 |
18 years ago |
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Russell Bryant
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a36833e3c2 |
Update dundi.conf to indicate that the asterisk.conf entityid option can be used
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 |
18 years ago |
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Tilghman Lesher
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9471b87d27 |
Merge the adaptive realtime branch, which will make adding new required fields
to realtime less painful in the future. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120789 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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76506b7baa |
Move compatibility options into asterisk.conf, default them to on for upgrades,
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 |
18 years ago |
Joshua Colp
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e4d1b39bd8 |
Merged revisions 118646 via svnmerge from
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 |
18 years ago |
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Tilghman Lesher
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932fd1aa5f |
Merged revisions 118358 via svnmerge from
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 |
18 years ago |
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Tilghman Lesher
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9276a4370c |
Add a compatibility option for upgrading realtime extensions
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117986 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Sean Bright
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3d412a7bb3 |
Minor text fix. roster -> resource.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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fced823c08 |
Change the default for the pridialplan parameter to the far more common case of
'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 |
18 years ago |
Luigi Rizzo
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f0093bfc42 |
fix example configuration for video support in chan_oss
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Jason Parker
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424a7816ea |
Merged revisions 116409 via svnmerge from
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 |
18 years ago |
Claude Patry
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485b1d9be1 |
fix a sample since we now required , and not | for the arguments separator
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115595 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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8b1d52c9a5 |
Allow a password change to be validated by an external script.
(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 |
18 years ago |
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Joshua Colp
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f4237076bf |
Add support for specifying the registration expiry on a per registration 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 |
18 years ago |
Mark Michelson
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e37dafdd3a |
Adding new configuration options to app_queue. This adds two new values
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 |
18 years ago |
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Joshua Colp
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1e066813ac |
Add support for authenticating on a NOTIFY request. This is useful for 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 |
18 years ago |
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Jeff Peeler
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41fd7a6a21 |
(closes issue #6113)
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 |
18 years ago |
Steve Murphy
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5fb4b1bbe5 |
This is the scariest commit I've done in a long time. This is the 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 |
18 years ago |
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Tilghman Lesher
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0dd46a6bf0 |
Make the sample config match the contributed LDAP schema
(Closes issue #12421) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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ded5ec5b5d |
Merged revisions 113874 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines If the [csv] section does not exist in cdr.conf, then an unload/load sequence is needed to correct the problem. Track whether the load succeeded with a variable, so we can fix this with a simple reload event, instead. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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137c02a020 |
Permit message wrap-around during message retrieval.
(closes issue #12254) Reported by: andrew Patches: bug-12253.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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36cd3d0107 |
Additional note
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113245 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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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 |
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Jason Parker
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63f574ceb4 |
Merged revisions 113118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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c6453ded22 |
Update sample configurations to make virtual hosting more obvious.
(closes issue #11969) Reported by: pprindeville Patches: acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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7741ed8bcc |
Update the sample configuration, to use Macro less (since it's now deprecated).
(closes issue #12293) Reported by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by pprindeville (license 347) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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738e4ec94e |
Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue #9239) Reported by: sunder git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110631 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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a567b41083 |
Note that the TCP and TLS support is currently considered experimental and
is subject to change while we work out the remaining issues. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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58fa8e6e9e |
Change back to using ldap_initialize() and let the user specify a URL directly,
instead of trying to piece it together, badly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Mark Michelson
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cd7efcf4e7 |
Add option 'randomperiodicannounce' to queues.conf. Setting this will
allow the list of periodic announcments specified to be played in a random order instead of being played sequentially. (closes issue #6681) Reported by: alt_phil Tested by: putnopvut git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Olle Johansson
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0de4eba640 |
Add manager peerstatus events when peer can't authenticate.
(closes issue #11959) Reported by: mostyn Patches: peerstatus3.patch uploaded by mostyn (license 398) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Jason Parker
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93b0f037b4 |
Add sample events for aastra phones.
aastra-check-cfg is the same as the other check-cfg entries, and aastra-xml is to load a pre-configured xml script. (closes issue #12229) Reported by: gowen72 Patches: aastra.patch uploaded by gowen72 (license 432) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109111 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
Kevin P. Fleming
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a3a8aa6547 |
add support for named sections in zapata.conf, and fix a few bugs in config file parsing
(closes issue #9503) Reported by: tzafrir Patches: fix_cleanups uploaded by tzafrir (license 46) zapata_sections uploaded by tzafrir (license 46) skipchannel_options uploaded by tzafrir (license 46) conf_sample uploaded by tzafrir (license 46) patches updated by me to better conform to coding guidelines and fix some problems git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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0b97554307 |
Add contributed script for separation of database access from Asterisk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@107721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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8a411ccf83 |
Create a centralized configuration option for silencethreshold
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Joshua Colp
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7422f0ee37 |
Add documentation for setting username/password in SIP dial string.
(closes issue #11587) Reported by: sobomax Patches: dialstring_doc.diff uploaded by sobomax (license 359) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Tilghman Lesher
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4aff24881b |
Bring Voicetronix driver up to date with current drivers
(closes issue #12084) Reported by: mmickan Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) module.h.diff uploaded by mmickan (license 400) vpb.conf.sample uploaded by mmickan (license 400) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Russell Bryant
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3a8756c9b4 |
Merged revisions 104119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines Merge changes from team/russell/smdi-1.4 This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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Brett Bryant
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55aaa80d15 |
Adding more tls configuration details to sip.conf sample, with a list of valid ciphers provided in both files. .. First commit since July, woot
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |