Age | Commit message (Collapse) | Author | Files | Lines |
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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: http://svn.digium.com/svn/asterisk/trunk@135126 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@135067 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@134401 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134086 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines
Fixes for AST-2008-010 and AST-2008-011
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@132643 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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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: http://svn.digium.com/svn/asterisk/trunk@130951 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130040 f38db490-d61c-443f-a65b-d21fe96a405b
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in the documentation.
Patch provided by caio1982 (license 22)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128599 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@128525 f38db490-d61c-443f-a65b-d21fe96a405b
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- 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: http://svn.digium.com/svn/asterisk/trunk@128524 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128237 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128236 f38db490-d61c-443f-a65b-d21fe96a405b
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application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128122 f38db490-d61c-443f-a65b-d21fe96a405b
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explanation of the change may be found in configs/queues.conf.sample
(closes issue #12690)
Reported by: atis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127720 f38db490-d61c-443f-a65b-d21fe96a405b
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acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127558 f38db490-d61c-443f-a65b-d21fe96a405b
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without it being defined by each sip user.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127154 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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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: http://svn.digium.com/svn/asterisk/trunk@125891 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) | 4 lines
Document ackcall=always.
(closes issue #12852)
Reported by: davidw
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125223 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12838, related to issue #12198)
Reported by: pabelanger
Patches:
http.conf.diff2 uploaded by pabelanger (license 224)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@125191 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124669 f38db490-d61c-443f-a65b-d21fe96a405b
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(And this time, do it in the correct repository :-))
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124635 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12830)
Reported by: jcollie
Patches:
astertisk-trunk-121496-alternate-extensions.patch uploaded by jcollie (license 412)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124049 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123887 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122977 f38db490-d61c-443f-a65b-d21fe96a405b
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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122234 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@121559 f38db490-d61c-443f-a65b-d21fe96a405b
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to set the entityid used in DUNDi, as well.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121441 f38db490-d61c-443f-a65b-d21fe96a405b
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to realtime less painful in the future.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120789 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@120171 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118647 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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'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: http://svn.digium.com/svn/asterisk/trunk@117182 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@117053 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | 1 line
Document exitcontext in app_voicemail sample config
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(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: http://svn.digium.com/svn/asterisk/trunk@115582 f38db490-d61c-443f-a65b-d21fe96a405b
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basis in the register line. This comes from a Switchvox patch. (issue AST-24)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114912 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@114906 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@114529 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@114190 f38db490-d61c-443f-a65b-d21fe96a405b
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(Closes issue #12421)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114088 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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.
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(closes issue #12254)
Reported by: andrew
Patches:
bug-12253.diff uploaded by snuffy (license 35)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@113731 f38db490-d61c-443f-a65b-d21fe96a405b
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