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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r226531 | file | 2009-10-29 15:11:26 -0300 (Thu, 29 Oct 2009) | 6 lines
Add an option to enabling passing music on hold start and stop requests through instead of
acting on them in chan_local.
(closes issue #14709)
Reported by: dimas
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r226377 | lmadsen | 2009-10-28 14:48:29 -0500 (Wed, 28 Oct 2009) | 7 lines
Update CALLINGSUBADDR channel variable documentation.
(closes issue #15734)
Reported by: alecdavis
Patches:
channelvariables.tex.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
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SIPShowPeer AMI action.
(closes issue #15990)
Reported by: _brent_
Patches:
sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388)
Review: https://reviewboard.asterisk.org/r/381/
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For more details:
http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/
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r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009) | 11 lines
Update imapstorage.txt documentation.
Updated the imapstorage.txt documentation to reflect that issues with
c-client versions older than 2007 seem to cause crashing issues that
are not seen with more recent versions. Documentation has been updated
to reflect this.
(closes issue #14496)
Reported by: vbcrlfuser
Patches:
__20090727-imap-documentation-patch.txt uploaded by lmadsen (license 10)
Tested by: lmadsen, mmichelson, dbrooks
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It is clear from multiple mailing list, forum, wiki and other sorts of posts
that users don't really understand the effects that the 'canreinvite' config
option actually has, and that in some cases they think that setting it to 'no'
will actually cause various other features (T.38, MOH, etc.) to not work properly,
when in fact this is not the case. This patch changes the proper name of the
option to what it should have been from the beginning ('directmedia'), but
preserves backwards compatibility for existing configurations.
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(closes issue #15516)
Reported by: snuffy
Patches:
bug_odbc_tex_update_v2.diff uploaded by snuffy (license 35)
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This commit introduces the security events API. This API is to be used by
Asterisk components to report events that have security implications.
A simple example is when a connection is made but fails authentication. These
events can be used by external tools manipulate firewall rules or something
similar after detecting unusual activity based on security events.
Inside of Asterisk, the events go through the ast_event API. This means that
they have a binary encoding, and it is easy to write code to subscribe to these
events and do something with them.
One module is provided that is a subscriber to these events - res_security_log.
This module turns security events into a parseable text format and sends them
to the "security" logger level. Using logger.conf, these log entries may be
sent to a file, or to syslog.
One service, AMI, has been fully updated for reporting security events.
AMI was chosen as it was a fairly straight forward service to convert.
The next target will be chan_sip. That will be more complicated and will
be done as its own project as the next phase of security events work.
For more information on the security events framework, see the documentation
generated from doc/tex/. "make asterisk.pdf"
Review: https://reviewboard.asterisk.org/r/273/
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Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
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When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
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I still need to set up tests to make sure my examples are completely correct,
but I ran out of time tonight and felt that they at least would give an idea as
to how to use calendaring. I will try to test the examples and do some cleanup
on the docs tomorrow night.
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This would allow for one to add a caller to a specific place in the
queue instead of just placing the caller in the back every time. To help
facilitate some interesting manipulations, a new channel variable called
QUEUEPOSITION has been added. When a caller is removed from a queue, his
position in that queue is stored in the QUEUEPOSITION variable. One such
strategy an administrator can employ is to allow for the removal of a caller
from one queue followed by the insertion of the same caller into a separate
queue in the same position.
Review: http://reviewboard.digium.com/r/189
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a dialplan variable.
This adds a dialplan variable (SIP_CODEC_OUTBOUND) which controls
the codec offered for an outgoing SIP call. This is much like the
SIP_CODEC dialplan variable and has the same restrictions. The codec
set must be one that is configured for the call.
(closes issue #13243)
Reported by: samdell3
Patches:
13243.diff uploaded by file (license 11)
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(closes issue #14566)
Reported by: klaus3000
Patches:
ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65)
ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65)
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separate file instead of the version controlled one.
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(closes issue #14049)
Reported by: kshumard
Patches:
doc.tex.qos.tex.patch uploaded by kshumard (license 92)
(Slight modifications by seanbright)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162659 | mmichelson | 2008-12-10 10:10:25 -0600 (Wed, 10 Dec 2008) | 8 lines
Add missing documentation to misdn.txt
(closes issue #14052)
Reported by: festr
Patches:
misdn.txt.patch uploaded by festr (license 443)
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This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
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Setting _SIPFROMDOMAIN in a channel will set the domain we use for the URI in the outbound call leg.
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more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
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PRIVACYMRGSTATUS variable
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so people can do pattern matching on the input
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, 13 Aug 2008) | 1 line
Update docs to reflect the change to cdr_tds
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the user manual stuff in doc/tex/.
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(closes issue #13211)
Reported by: eliel
Patches:
fixtexerrors.patch uploaded by eliel (license 64)
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respectively. Also, take the opportunity to clean up the CLI prompt
generation code.
(closes issue #13175)
Reported by: eliel
Patches:
cliprompt.patch uploaded by eliel (license 64)
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To make sure nobody commits script-modified files we first make a backup
of asterisk.tex, run the script, generate the pdf and / or html,
and put the original asterisk.tex back.
This will guard us for the stuff that happened before that someone committed
a locally modified asterisk.tex, with changes done by this script.
(closes issue #13062)
Reported by: mvanbaak
Patches:
sed_without-i-v3.diff uploaded by mvanbaak (license 7)
Tested by: mvanbaak
Feedback from Corydon. Thanks for taking the time to go through this.
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This got lost in commit 97634
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in the documentation.
Patch provided by caio1982 (license 22)
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FreeTDS 0.60. Update the docs to reflect that we can now compile and run
against all modern releases of FreeTDS (0.60 through 0.82)
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