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r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di, 03 Apr 2007) | 15 lines
Merged revisions 59788,59803 via svnmerge from
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r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 lines
Use the new sysfs way of mISDN 1.2 to check if a port is NT or not.
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r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, 03 Apr 2007) | 2 lines
ptp is the 5th bit, not the 4th.
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probably others, too. I don't really have time to work on it at the moment,
so I am just going to revert it for now.
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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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just having one that can be re-used. There is no functional change here (that
is intentional, anyway!).
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r59363 | russell | 2007-03-29 12:43:52 -0500 (Thu, 29 Mar 2007) | 6 lines
When building a response to a subscription, the "from" must be the full Jabber
ID. This fixes some problems where jabber users are not able to add their
Asterisk account to their user list, since they are unable to get Asterisk
to approve their subscription. (issue #8210, reported by caspy, and verified
by bradtem)
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r59361 | file | 2007-03-29 13:38:55 -0400 (Thu, 29 Mar 2007) | 10 lines
Merged revisions 59360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 lines
Keep a global array of variables indicating whether certain conference rooms are in use. This ensures that two people going into a new dynamic conference when the 'e' option is set don't go into the same conference room. (issue #8835 reported by eliel)
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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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r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) | 8 lines
When the IAX2 read callback gets called, return NULL instead of a "null frame".
This will cause Asterisk to hangup the call instead of keep trying whatever it
was doing. Under normal conditions, this function would *never* be called.
However, the author of this patch says an error will occur that will cause it
to get called every 100 thousand calls or so. When this does happen, it puts
the channel in a loop that eventually brings down the system. So, hangup up
the call is certainly a better alternative. (issue #8286, john)
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r59261 | murf | 2007-03-27 12:16:32 -0600 (Tue, 27 Mar 2007) | 1 line
via 9373 (duplicate context in AEL crashes asterisk), kpfleming pointed on asterisk-dev, that DECLINE in this case the proper thing to do. This change now has it doing the proper thing.
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r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines
Convert the RTPQOS function to just be additional parameter of the CHANNEL
function. This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
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r59228 | murf | 2007-03-26 15:41:32 -0600 (Mon, 26 Mar 2007) | 1 line
fix for 9373 (duplicate context in AEL crashes asterisk). I turned a duplicate context from a WARNING to an ERROR. Now you get a module load failure, and asterisk just exits. That's better than a crash, right\?
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r59230 | tilghman | 2007-03-26 16:45:44 -0500 (Mon, 26 Mar 2007) | 2 lines
Oops, this should be case insensitive
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r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)
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r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines
* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
* add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in'
(the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected).
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Russell Bryant
Nadi Sarrar
Joshua Colp
Steve Murphy
Jason Parker
Matthew Fredrickson
Tilghman Lesher
Dwayne M. Hubbard