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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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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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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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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r60521 | russell | 2007-04-06 13:58:46 -0500 (Fri, 06 Apr 2007) | 16 lines
Fix a few problems with SLA. (issue #9459, reported by francesco_r, fixed by me)
* The original behavior was that if one station put a call on hold, another one
picked it up, and then hung up, the code would still consider the call on
hold by the first station, so the trunk would not be hung up. However, to
better comply with what most people seem to expect it to behave, it will now
hang up the trunk.
* Fix a problem with "barge=no". This was only intended to prevent people from
joining calls that are in progress. However, it also prevented other people
from picking up a call that was on hold. This has been fixed.
* When there are no active stations on a trunk and it is on hold, the code now
indicates the HOLD and UNHOLD conditions to the trunk channel. This allows
music on hold to be played to the trunk when it is on hold.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Dwayne M. Hubbard
Russell Bryant
Nadi Sarrar
Steve Murphy
Tilghman Lesher
Doug Bailey
Olle Johansson
Kevin P. Fleming
Jason Parker