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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r142865 | tilghman | 2008-09-12 15:37:18 -0500 (Fri, 12 Sep 2008) | 11 lines
Create rules for disallowing contacts at certain addresses, which may
improve the security of various installations. As this does not change
any default behavior, it is not classified as a direct security fix for
anything within Asterisk, but may help PBX admins better secure their
SIP servers.
(closes issue #11776)
Reported by: ibc
Patches:
20080829__bug11776.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, blitzrage
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explanation to the file, since that additional text helps people understand
the concept.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #13411)
Reported by: caio1982
Patches:
res_jabber_autoprune1.diff uploaded by caio1982 (license 22)
Tested by: caio1982
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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existing names for backwards compatibility).
(closes issue #13370)
Reported by: jsturtevant
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Reported by: erousseau
This was a reasonable enhancement request, which was
easy to implement. Since it's an enhancement, it
could only be applied to trunk.
Basically, for accounting where "initiated" seconds
are billed for, if the microseconds field on the end
time is greater than the microseconds field for the
answer time, add one second to the billsec field.
The implementation was requested by erousseau, and
I've implemented it as requested. I've updated the
CHANGES, the cdr.conf.sample, and the .h files
accordingly, to accept and set a flag for the
corresponding new option. cdr.c adds in the extra
second based on the usec fields if the option is
set. Tested, seems to be working fine.
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* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/tex/misdn.tex
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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and confusing code pieces. Clarify the logic within
queues.conf.sample.
(closes issue #12690)
Reported by: atis
Patches:
queue_timeoutpriority.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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the port configuration option from cdr_tds.conf. So go ahead and
remove it from the sample config.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138442 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines
More fixes for realtime peers.
(closes issue #12921)
Reported by: Nuitari
Patches:
20080804__bug12921.diff.txt uploaded by Corydon76 (license 14)
20080815__bug12921.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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(closes issue #13314)
Reported by: kue
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines
Comments in this config file were aligned only if your tab size was set to 8.
So, convert tabs to spaces so that things should be aligned regardless of what
tab size you use in your editor.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) | 5 lines
* The allowed_bearers setting in misdn.conf misspelled one
of its options: digital_restricted.
* Fixed some other spelling errors and typos.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) | 2 lines
fix a config sample typo
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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
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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
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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: https://origsvn.digium.com/svn/asterisk/trunk@135067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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
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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.
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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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in the documentation.
Patch provided by caio1982 (license 22)
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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)
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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
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application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset
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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: https://origsvn.digium.com/svn/asterisk/trunk@127720 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
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without it being defined by each sip user.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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)
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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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(closes issue #12838, related to issue #12198)
Reported by: pabelanger
Patches:
http.conf.diff2 uploaded by pabelanger (license 224)
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(And this time, do it in the correct repository :-))
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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: https://origsvn.digium.com/svn/asterisk/trunk@124049 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
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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
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to set the entityid used in DUNDi, as well.
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to realtime less painful in the future.
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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: https://origsvn.digium.com/svn/asterisk/trunk@120171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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