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r62299 | russell | 2007-04-28 16:56:20 -0500 (Sat, 28 Apr 2007) | 2 lines
Note that the "talker optimization" option will be enabled by default in 1.6
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62300 f38db490-d61c-443f-a65b-d21fe96a405b
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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This introduces two new dialplan functions: DUNDIQUERY and DUNDIRESULT.
DUNDIQUERY lets you intitiate a DUNDi query from the dialplan. Then,
DUNDIRESULT will let you find out how many results there are, and access each
one without having to the query again.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62267 f38db490-d61c-443f-a65b-d21fe96a405b
minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604, patch by Matthew Roth)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines
Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks. This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel. However, this is the
wrong thing to do. It should *always* return 0, instead. When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r62171 | russell | 2007-04-27 11:14:11 -0500 (Fri, 27 Apr 2007) | 6 lines
If no variables were passed into pbx_substitute_variables_helper_full(), then
don't even bother creating a temporary bogus channel, since that is only for
allowing certain functions to operate on the variables as if they were on a
channel. Most importantly, this fixes a crash.
(issue #9613, reported by callguy, fixed by me)
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r61787 | russell | 2007-04-24 16:34:53 -0500 (Tue, 24 Apr 2007) | 12 lines
Merged revisions 61786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61786 | russell | 2007-04-24 16:33:59 -0500 (Tue, 24 Apr 2007) | 4 lines
Don't crash if a manager connection provides a username that exists in
manager.conf but does not have a password, and also requests MD5
authentication. (ASA-2007-012)
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r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines
Improve DTMF handling in ast_read() even more in response to a discussion on
the asterisk-dev mailing list. I changed the enforced minimum length of a
digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in
between digits. These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.
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r61774 | russell | 2007-04-24 11:16:41 -0500 (Tue, 24 Apr 2007) | 5 lines
Add a few more state changes in handle_frame_ownerless() so that the SLA code
will get notified of these changes even when an owner channel is not provided.
This isn't from a specific bug report, it's just something I noticed while
poking around.
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r61765 | russell | 2007-04-23 13:17:00 -0500 (Mon, 23 Apr 2007) | 5 lines
Some dialplan functions, such as CUT(), expect to operate on variables on a
channel. So, this little hack lets them work in places where a channel doesn't
exist, such as within DUNDi configuration.
(issue #9465, reported and patched by Corydon76, testing by blitzrage)
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r61763 | russell | 2007-04-23 12:57:32 -0500 (Mon, 23 Apr 2007) | 4 lines
Ensure that digits passing through Asterisk have a reasonable minimum length.
It is currently 100 ms. If someone thinks this should be different, feel free
to speak up. (related to issues #8944, #9250, and #9348)
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r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines
Avoid invalid seqno cycling detection.
Per comment from Dave Troy:
This adds back in some simple typecasting I had in an earlier version
which I realize now may be breaking things.
Issue #9554.
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This set of changes adds OSP support to chan_iax2. However, I have modified
the patch a bit from what was submitted. You now use the CHANNEL() function
to get and set the OSP token for IAX2.
(issue #8531, reported by and original patch by homesick, patch updated by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61694 | qwell | 2007-04-20 14:51:49 -0500 (Fri, 20 Apr 2007) | 13 lines
Merged revisions 61692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61692 | qwell | 2007-04-20 14:49:54 -0500 (Fri, 20 Apr 2007) | 5 lines
If the '* to hangup' option is not enabled, we don't need to disable * as a valid exit key.
If it was enabled, this statement would've never been checked in the first place.
Issue #9552
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r61681 | tilghman | 2007-04-18 21:45:05 -0500 (Wed, 18 Apr 2007) | 13 lines
Merged revisions 61680 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61680 | tilghman | 2007-04-18 21:30:18 -0500 (Wed, 18 Apr 2007) | 5 lines
Bug 9557 - Specifying the GetVar AMI action without a Channel parameter can
cause Asterisk to crash. The reason this needs to be fixed in the functions
instead of in AMI is because Channel can legitimately be NULL, such as when
retrieving global variables.
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blocks instead of one large monolithic app. Supports multiple templates
and is designed mostly for voicemail delivery over e-mail.
There's a todo with a list of ideas in the source code if you want
to contribute. Feedback is appreciated!
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