https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61477 | russell | 2007-04-11 11:05:29 -0500 (Wed, 11 Apr 2007) | 13 lines
Merged revisions 61476 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines
If someone sets the "useragent" option in sip.conf to be empty, then don't add
the User-Agent header at all. It is an optional header, anyway. Also, the bug
report says that some of Japan's SIP providers don't allow it for some weird
reason. (issue #9488, reported by makoto, fixed by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines
Merged revisions 61426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines
Fix a bug with switching between host=dynamic and using specific hosts for
peers. The code would only reset the peer's address when it is dynamic if
it was a new peer structure. Now, it will also reset the address if it was
already in the peer list, but before the reload, it was not dynamic.
(issue #9515, reported by caio1982, fixed by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines
Merged revisions 61376 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines
Remove the attempt at reporting configuration errors in sip.conf. This can
cause a bunch of improper messages when using realtime. I give up. As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r61220 | russell | 2007-04-10 11:05:55 -0500 (Tue, 10 Apr 2007) | 5 lines
File upload support was added to solve some needs for the Asterisk GUI.
However, after much discussion, it has been decided that adding this to 1.4 is
not in the best interests of the project. It has been removed here, but will
remain in trunk.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r61042 | russell | 2007-04-09 14:40:29 -0500 (Mon, 09 Apr 2007) | 2 lines
Remove various files that I thought I already removed.
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r61044 | russell | 2007-04-09 14:41:04 -0500 (Mon, 09 Apr 2007) | 2 lines
Remove another directory that should no longer be there
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
"ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more
in line with "core show channeltypes"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
