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configuration issues
(closes issue #12151)
Reported by: caio1982
Patches:
DB_metric3.diff uploaded by caio1982 (license 22)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106186 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103725 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103711 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103705 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100679 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@100422 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99647 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98988 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #11763)
Reported by: IgorG
Patches:
docupd.v1.diff uploaded by IgorG (license 20)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98695 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@98454 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97634 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96934 f38db490-d61c-443f-a65b-d21fe96a405b
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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).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@94370 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89451 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89417 f38db490-d61c-443f-a65b-d21fe96a405b
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greetingsfolder options
in voicemail.conf
(closes issue #11133, reported by selsky, patched by blitzrage)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89075 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89069 f38db490-d61c-443f-a65b-d21fe96a405b
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SVN-group-chan_unistim-r88326M-/trunk
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88327 f38db490-d61c-443f-a65b-d21fe96a405b
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Remove old unused defines for old style.
Closes issue 10860, patch by IgorG.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85764 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85683 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85547 f38db490-d61c-443f-a65b-d21fe96a405b
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Here it is.
(closes issue #10962)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85539 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85526 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85519 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85097 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84580 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81233 f38db490-d61c-443f-a65b-d21fe96a405b
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AgentConnect
Closes issue #10349, patch by eliel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77879 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77432 f38db490-d61c-443f-a65b-d21fe96a405b
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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)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77284 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76874 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@73449 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72986 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72982 f38db490-d61c-443f-a65b-d21fe96a405b
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