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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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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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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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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I started this for use with SLA but ended up deciding not to use it. However,
there is no reason not to put this part in, anyway.
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r60850 | tilghman | 2007-04-08 22:01:12 -0500 (Sun, 08 Apr 2007) | 10 lines
Merged revisions 60849 via svnmerge from
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r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines
Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list).
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r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines
To be able to achieve the things that we would like to achieve with the
Asterisk GUI project, we need a fully functional HTTP interface with access
to the Asterisk manager interface. One of the things that was intended to be
a part of this system, but was never actually implemented, was the ability for
the GUI to be able to upload files to Asterisk. So, this commit adds this in
the most minimally invasive way that we could come up with.
A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to check
permissions of active manager sessions was added by Dwayne Hubbard. Then,
hacking this all together and do doing the modifications necessary to the HTTP
interface was done by me.
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r60137 | russell | 2007-04-04 12:40:10 -0500 (Wed, 04 Apr 2007) | 14 lines
Merged revisions 60134 via svnmerge from
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r60134 | russell | 2007-04-04 12:38:47 -0500 (Wed, 04 Apr 2007) | 6 lines
It is valid to redirect channels via the manager interface that are not in the
UP state. Instead of checking for that to prevent to ensure a dead channel
doesn't get redirected, just use the ast_check_hangup() API call.
(issue #9457, reported by Callmewind, patch by me)
(related to issue #8977)
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r59887 | russell | 2007-04-03 13:01:49 -0500 (Tue, 03 Apr 2007) | 13 lines
Merged revisions 59886 via svnmerge from
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r59886 | russell | 2007-04-03 12:58:19 -0500 (Tue, 03 Apr 2007) | 5 lines
When doing a built-in blind or attended transfer, restore the ability to use '#'
to terminate the number and immediately do the transfer instead of having to
dial the number and just wait for the feature digit timeout.
(issue #8366, xueliangliang)
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r59654 | russell | 2007-04-02 10:39:07 -0500 (Mon, 02 Apr 2007) | 14 lines
Merged revisions 59608 via svnmerge from
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r59608 | russell | 2007-04-01 17:35:25 -0500 (Sun, 01 Apr 2007) | 6 lines
Add the SO_REUSEADDR flag to sockets handled by netsock. This is needed by
the patch that went in for issue 7874. chan_iax2 needs to be able to create
socket that is lisetning on INADDR_ANY, but also be able to bind sockets to
specific addresses. (Thanks to Stevenson on the asterisk-dev mailing list
for explaining why this flag was needed.)
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r59358 | russell | 2007-03-29 12:17:41 -0500 (Thu, 29 Mar 2007) | 13 lines
Merged revisions 59357 via svnmerge from
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r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines
If an error occurs when reading from an RTP socket, and the error code does not
indicate that we should try again, then return NULL instead of a "null frame".
This will prevent Asterisk from trying over and over again, and eventually
causing the system to crash. (issue #8285, john)
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