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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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include a bunch of empty entries.
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version of the c-client library. (issue #10047, jcollie)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r71519 | russell | 2007-06-25 07:40:06 -0500 (Mon, 25 Jun 2007) | 2 lines
Fix a typo in the Asterisk mib. (issue #10048, Matti)
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allowing users to declare local variables.
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with carriers that do Polarity/DTMF CID signalling.
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via SNMP.
(closes issue #9637, initial patch from jeffg, modified by me)
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members example. (issue #9813 reported by Mike Anikienko)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r66821 | tilghman | 2007-05-31 18:03:28 -0500 (Thu, 31 May 2007) | 2 lines
Issue 9850 - update preferred command line syntax
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r66398 | file | 2007-05-29 11:43:16 -0400 (Tue, 29 May 2007) | 2 lines
Update datastores documentation. (issue #9801 reported by mnicholson)
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triggered the upgrade.
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not properly added. Thanks to IgorG for pointing it out! :)
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entries in the queue log.
(issue #7561, reported and originally patched by fkasumovic, patch slightly
modified and updated to trunk by me)
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file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)
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and DUNDIRESULT. Also, update the automatically generated application docs.
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doc/asterisk-conf.tex
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(issue #9416, phsultan)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r58946 | tilghman | 2007-03-15 18:52:48 -0500 (Thu, 15 Mar 2007) | 2 lines
Refashion dump command to match common syntax and update the resulting appdocs TeX file
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r58941 | russell | 2007-03-15 18:24:09 -0500 (Thu, 15 Mar 2007) | 1 line
add a link to the rubber homepage
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r58937 | russell | 2007-03-15 17:51:29 -0500 (Thu, 15 Mar 2007) | 2 lines
Add Asterisk version information to the generated PDF
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | 13 lines
Merge changes from svn/asterisk/team/russell/LaTeX_docs.
* Convert most of the doc directory into a single LaTeX formatted document
so that we can generate a PDF, HTML, or other formats from this
information.
* Add a CLI command to dump the application documentation into LaTeX format
which will only be include if the configure script is run with
--enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included. However, you
can simply run "make asterisk.pdf" to generate it yourself. We may include
it in release tarballs or have automatically generated ones on the web site,
but that has yet to be decided.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r58902 | russell | 2007-03-14 12:04:38 -0500 (Wed, 14 Mar 2007) | 2 lines
Add a more basic example setup to the examples section
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r58897 | russell | 2007-03-14 11:40:22 -0500 (Wed, 14 Mar 2007) | 11 lines
Merged revisions 58896 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58896 | russell | 2007-03-14 11:38:48 -0500 (Wed, 14 Mar 2007) | 3 lines
Add a note to the security file that the Asterisk CLI and log files may contain
sensitive information, and that people should keep this in mind.
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* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a
SQLite3 database. (issue #7149, alerios)
* Add new module, res_config_sqlite, which adds realtime database configuration
support for SQLite version 2. I decided that this was ok since we didn't have
any realtime support for version 3. If someone ports this to version 3, then
version 2 support can be removed or marked deprecated.
(issue #7790, rbarun_proformatique)
* Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom.
Also, note that there were other modules on the bug tracker that did not make
the cut because they provided some duplicated functionality. Those are:
* cdr_sqlite3 (issue #6754, moy)
* cdr_sqlite3 (issue #8694, bsd)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r58638 | russell | 2007-03-09 17:59:10 -0600 (Fri, 09 Mar 2007) | 8 lines
Merge some updates to the SLA documentation. I plan to keep working on this
to explain all of the expected behavior with call handling, configuration
details for specific phones, and other things. However, I got tired of doing
it in plain text, so I switched to using LaTeX. I have included the PDF version.
I haven't been able to get a nice looking plain text version out of it yet, but
I'm not terribly concerned since this is supposed to be more of the manual,
while the plain text sample configuration file is the reference.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines
Merge changes from svn/asterisk/team/russell/sla_updates
* Originally, I put in the documentation that only Zap interfaces would be
supported on the trunk side. However, after a discussion with Qwell, we came
up with a way to make IP trunks work as well, using some things already in
Asterisk. So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
in SLA. The station's channel needs to be passed to the dial API when
dialing the trunk.
* Change a WARNING message to DEBUG in channel.h. This message is of no use
to users.
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r57207 | russell | 2007-02-28 17:01:52 -0600 (Wed, 28 Feb 2007) | 2 lines
minor tweaks to the sla docs
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r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) | 18 lines
Merge changes from team/russell/sla_updates.
This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
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realtime family.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines
Merge team/russell/sla_rewrite
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
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r53715 | kpfleming | 2007-02-09 10:42:22 -0600 (Fri, 09 Feb 2007) | 2 lines
clarify the fact that voicemail IMAP storage cannot be built against a distro's binary c-client library package (at least not at this time)
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