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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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Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
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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.
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stop of monitoring (one-touch monitor).
Keep messages short, since the other party is waiting while one party hear the message...
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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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configuration issues
(closes issue #12151)
Reported by: caio1982
Patches:
DB_metric3.diff uploaded by caio1982 (license 22)
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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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(closes issue #11763)
Reported by: IgorG
Patches:
docupd.v1.diff uploaded by IgorG (license 20)
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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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go a long way towards preventing unexplainable hangs experienced by people. In the
case of MWI hangs, this also will mean that the SIP port isn't blocked anymore.
(closes issue #11665, reported by yehavi)
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adjust
the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending
on the amount of time passed. The purpose is to allow the call to open up to more (or maybe
just different) members without the caller's losing his place in the queue. See
configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample
for how to associate a rule with a queue.
Along with the functional changes, new CLI and manager commands exist to show the rules defined and
there is an additional CLI command to reload the queue rules.
Future enhancements that may be made: support for realtime queue rules and support for dynamically adding
a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write
this myself very soon).
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- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1 line
closes issue #11324; break statements missing in switch cases.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov 2007) | 4 lines
Clarify documentation a bit, include that a frame has to pass through the core in order for the Local channel optimization to happen.
(closes issue #11246)
Reported by: jon
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greetingsfolder options
in voicemail.conf
(closes issue #11133, reported by selsky, patched by blitzrage)
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much identical to the S() and L() options to Dial(). They let you set
timeouts for the conference, as well as have warning sounds played to
let the caller know how much time is left, and when it is running out.
(closes issue #8030)
Reported by: areski
Patches:
meetme_timeout_timelimit_v2.patch uploaded by areski (license 29)
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SVN-group-chan_unistim-r88326M-/trunk
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Remove old unused defines for old style.
Closes issue 10860, patch by IgorG.
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- Many uses of the astlisting environment around verbatim text to ensure that
it gets properly formatted and doesn't run off the page.
- Update some things that have been deprecated.
- Add escaping as needed
- and more ...
(closes issue #10978)
Reported by: IgorG
Patches:
texdoc-85542-1.patch uploaded by IgorG (license 20)
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Here it is.
(closes issue #10962)
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- Added development section (backtrace.tex)
- Correct filesystem path formating
- Replace all "|" argument separator to ","
- Endless count of spaces at the end of line
- Using astlisting to make listings do not take so much place
- Take back ASTRISKVERSION on first page
- Make localchannel.tex readable by inserting extra end of lines
(closes issue #10962)
Reported by: IgorG
Patches:
texdoc-85177-1.patch uploaded by IgorG (license 20)
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in the Dial command. The 'j' option _must_ be used in conjunction with the 'n'
option.
This feature will allow you to use the existing jitterbuffer implementation to
put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by
putting a local channel in the middle.
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Reported by: nic_bellamy
Patches:
2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by nic_bellamy (license 213)
Add support for configurable file locking methods. The default is "lockfile",
which is the old behavior. There is an additional option, "flock", which is
intended for use in situations where the lockfile method will not work, such as
with SMB/CIFS mounts.
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AgentConnect
Closes issue #10349, patch by eliel
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77424 | kpfleming | 2007-07-26 17:14:21 -0500 (Thu, 26 Jul 2007) | 2 lines
use new canonical name for download server
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r77429 | kpfleming | 2007-07-26 17:16:42 -0500 (Thu, 26 Jul 2007) | 2 lines
change protocol for downloads as well
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where verbatim text went off the end of the page on the PDF, and various
other improvements
(closes issue #10307, IgorG)
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numbers, I'm adding this update to round out (no pun intended) and make this FP-capable version of the Expr2 stuff interoperate better with previous integer-only usage, by providing Functions syntax, with 20 builtin functions for floating pt to integer conversions, and some general floating point math routines that might commonly be used also. Along with this, I made it so if a function was not a builtin, it will try and find it in the ast_custom_function list, and if found, execute it and collect the results. Thus, you can call system functions like CDR(), CHANNEL(), etc, from within $\[..\] exprs, without having to wrap them in $\{...\} (curly brace) notation. Did a valgrind on the standalone and made sure there's no mem leaks. Looks good. Updated the docs, too.
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for extracting application, function, manager, and agi documentation is the wrong
one to take. The most severe problem is that the output depends on which modules
are loaded as well as compile time options, which both determine which parts are
available.
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* Add a Makefile in doc/tex/ for generating PDF and HTML
* Add a README.txt file to doc/tex/ to document which tools are used and what
web sites to visit for getting them.
* Update build_tools/prep_tarball to put the proper Asterisk version string
in the automatically generated PDF for release tarballs
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