in trunk.
Merged revisions 128417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r128417 | oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines
Adding documentation on the T.140 support in Asterisk. This is a function that we're
the reference implementation on now. :-)
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r126226 | seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 lines
Merge in changes from my cdr-tds-conversion branch. This changes the internal
implementation from using the volatile libtds, to using the db-lib front end.
The unintended side effect of this is that we support (at least) versions 0.62
through 0.82 of the FreeTDS distribution without any #ifdef ugliness.
(closes issue #12844)
Reported by: jcollie
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r126513 | seanbright | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines
Cast a few more strings to char *, so that we can compile cleanly against
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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r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008) | 13 lines
Add a locking section to the coding guidelines document.
This section covers some locking fundamentals, as well as some information on
locking as it is used in Asterisk. It describes some of the ways that are used
and could be used to achieve deadlock avoidance. It also demonstrates the
unfortunate conclusion that with the use of recursive locks, none of the
constructs in use today are failsafe from deadlocks. Finally, it makes some
recommendations for new code being written. As proper locking strategies is a
complex subject, this section still has room for expansion and improvement.
This is a result of collaboration between Luigi Rizzo and myself on the
asterisk-dev mailing list.
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r111908 | russell | 2008-03-28 17:45:43 -0500 (Fri, 28 Mar 2008) | 3 lines
Note a minor race condition that I noticed while reviewing Jeff's changes
to this code.
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r111909 | russell | 2008-03-28 17:50:46 -0500 (Fri, 28 Mar 2008) | 3 lines
Make some notes about common usage of pbx_builtin_getvar_helper() that is not
thread-safe.
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that if we keep this in the tree, it will be much easier to keep up to date.
The page on asterisk.org just links to this on svn.digium.com/view
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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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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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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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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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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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