https://origsvn.digium.com/svn/asterisk/trunk
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r210564 | lmadsen | 2009-08-05 13:49:58 -0500 (Wed, 05 Aug 2009) | 19 lines
Merged revisions 210563 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/trunk
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r210190 | kpfleming | 2009-08-03 15:48:48 -0500 (Mon, 03 Aug 2009) | 11 lines
Rename 'canreinvite' option to 'directmedia', with backwards compatibility.
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@210191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
DUNDi uses a concept called the Entity ID for unique server identifiers. I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this. The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf. DUNDi will now use this global EID unless one is specified
in dundi.conf.
The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: atis
Tested by: murf
This upgrade adds the ~~ (concatenation) string operator to expr2.
While not needed in normal runtime pbx operation, it is needed when
raw exprs are being syntax checked. This plays into future syntax-
unification plans. By permission of atis, this addition in trunk
and the reason of why things are as they are will suffice to close
this bug.
I also added a short note about the previous addition of "sip show sched"
to the CLI in CHANGES, which I discovered I forgot in a previous commit.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb 2008) | 8 lines
Final round of changes for configure script logic for IMAP
Now if a directory is specified, then we will search that directory for
a source installation of the IMAP toolkit. If none is found, then we will
use that directory as the basis for detecting a package installation of
the IMAP c-client. If that check fails, then configure will fail.
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the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf. I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.
Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport. Tested on Linux and OS X.
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run "make asterisk.pdf" when not all of the right packages are installed.
(closes issue #10763)
Reported by: Corydon76
Patches:
20070919__bug10763.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
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