Age | Commit message (Collapse) | Author |
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option for discard of remote hold messages, and using the alternate logical channel mapping in Q.SIG instead of the default physical channel mapping.
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devices and lines.
(closes issue #13412)
Reported by: wedhorn
Patches:
config-restruct-v4.diff uploaded by wedhorn (license 30)
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when Asterisk was configured using --prefix and 'make samples' is run
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r149683 | bweschke | 2008-10-15 14:28:54 -0400 (Wed, 15 Oct 2008) | 4 lines
An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c
(closes issue #13709)
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(closes issue #12860)
Reported by: flyn
Patches:
res_ldap.conf.patch uploaded by flyn (license 503)
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the grammar, formatting, and introduce several clarifications to the text.
(Closes issue #13654)
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default config
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making the new/old message count available to local devices. (issue #AST-77)
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and "leavewhenempty" options are configured in queues.conf.
Instead of using vague terms like "yes," "no," "loose," and
"strict," we now accept a comma-separated list of values
to determine when to consider a member available.
Extended details can be found in the queues.conf.sample
file. Note also that the above four referenced values are
still accepted for backwards-compatibility, but are mapped
internally to the new method of representing the option.
AST-105
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reorganized sip_notify.conf.sample a bit as well. Tested snom
reboot on snom 360 and verified snom-check-cfg worked as well.
(closes issue #13601)
Reported by: mjc
Tested by: seanbright
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It was pretty sparsely documented.
This update fleshes out the pbx_lua module, to
add the switch statements to the extensions in the
extensions.lua file, as well as removing them when
the module is unloaded.
Many thanks to Matt Nicholson for his fine
contribution!
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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.
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(closes issue #13411)
Reported by: caio1982
Patches:
res_jabber_autoprune1.diff uploaded by caio1982 (license 22)
Tested by: caio1982
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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.
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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)
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the port configuration option from cdr_tds.conf. So go ahead and
remove it from the sample config.
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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
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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
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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)
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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)
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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)
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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
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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.
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