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now present that can not be used at the same time as chan_alsa or chan_oss.
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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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Reported by: eliel
Patch by: eliel
(Closes issue #11344)
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version 1.1 - hopefully a more consistent manager interface.
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"call" level.
(Closes issue #11015)
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"Username" still works, but is deprecated.
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- Restructure other changes to UPGRADE.txt and CHANGES
We're still looking for scripts that replace
asterisk -rx "show shannels concise"
by using the manager interface, but still produces the same output.
Anyone?
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Also, some cleanup to CHANGES.
Reported by: michael-fig
Patch by: michael-fig,tilghman
(Closes issue #11004)
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Closes issue #10614
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QUEUE_MEMBER_COUNT function.
A deprecation notice will be issued the first time QUEUE_MEMBER_COUNT is used.
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Remove old unused defines for old style.
Closes issue 10860, patch by IgorG.
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CLI-style output (closes issue #8254)
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issue #10398)
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redistributed/repartitioned to allow code re-use both inside and outside of Asterisk. This commit introduces the utils/conf2ael program, and an external config-file reader, for both normal config files, and for extensions.conf (context, exten, prio); It provides an API for programs outside of asterisk to use to play with the dialplan and config files.
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are now deprecated.
(Closes issue #10264)
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documentation on the
change to the Local channel name.
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applications
using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r72933 | murf | 2007-07-02 14:16:31 -0600 (Mon, 02 Jul 2007) | 1 line
support for floating point numbers added to ast_expr2 $\[...\] exprs. Fixes bug 9508, where the expr code fails with fp numbers. The MATH function returns fp numbers by default, so this fix is considered necessary.
........
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find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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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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behavior), this update makes the 't' option available, which calls ast_parse_device_state instead of ast_device_state. This option will not dive into the channel driver to find the status of the device (which could be good if sip devicestate isn't returning full status, for various reasons).
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2. Rename 'minmessage' to 'minsecs' for parity.
3. Make 'maxsecs' a per-user option, in addition to global.
(Issue # 8624)
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remove deprecated SetGlobalVar app
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the author originally intended for it
to function along with an option to keep backward compatible with "old-style" functionality in 1.2.
(#6595 - davetroy reported and patched w/some very minor mods/corrections)
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Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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application and
does not need special code in chan_sip any more.
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This means that if you're using this and upgrade to the revision where this was committed, you will need to update your table to the schema provided in doc/odbcstorage.txt
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40033 f38db490-d61c-443f-a65b-d21fe96a405b
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(Issue #7610)
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versions of Asterisk will still work properly
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