https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46780 | file | 2006-11-01 13:39:47 -0500 (Wed, 01 Nov 2006) | 2 lines
Force poll() emulation for Darwin to always be on. It's too broken to consider being used. This resolves the console issue OSX users have been seeing. I would have liked to autoconf this but I haven't been able to come up with a test case that works. Que sera.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
switch in the dialplan by setting the DUNDIDIALARGS channel variable.
(issue #8084, patch by bluecrow76, with small modifications and documentation
updates)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46778 | russell | 2006-11-01 13:26:35 -0500 (Wed, 01 Nov 2006) | 17 lines
Merged revisions 46776 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | 9 lines
soxmix and Asterisk expect different file extensions for certain formats. This
was already handled for the wav49 format. However, it was not handled for
ulaw and alaw. I fixed this in such a way that using the alternate extensions
for ulaw and alaw will only happen if we know we're calling soxmix, and not a
custom script defined using the MONITOR_EXEC variable. The wav49 processing
was left alone so that external scripts will see no behavior change.
(issue #7550, reported by mnicholson, proposed patch by junky, committed fix
is a bit different)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46775 | file | 2006-11-01 13:21:34 -0500 (Wed, 01 Nov 2006) | 2 lines
It's another round of chan_iax2 fixes! Should hopefully fix the deadlock issues people have been reporting. IAXtel now has qualify turned on for 800 peers and it is handling it fine.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Check the channel name string length to be zero, not non-zero
- Check the message string length to be zero, not non-zero
- unlock the channel *after* calling sendtext
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Might need better locking support, giving up if we can't get the lock. Right now,
using existing locking in find_resource
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46554 | russell | 2006-10-31 00:55:07 -0500 (Tue, 31 Oct 2006) | 5 lines
Add a small tweak to the code that checks to see whether destination formats
are translatable based on the source format. If we have already determined
that there is no translation path in one direction, don't bother checking the
other direction.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46526 | kpfleming | 2006-10-30 16:19:55 -0600 (Mon, 30 Oct 2006) | 3 lines
when unregistering a translator, don't rebuild the translation matrix unless needed
when filtering formats out of an offer, ensure we check for translation ability in both directions
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
various PBX installations and checks if a module is loaded before using
it.
example IFMODULE(chan_sip3.so)
issue #6671 in the bug tracker, finally gone. Thanks to mithraen for keeping
it updated.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
is to the person that configures asterisk. That we use it internally in the
contact header is a totally different story.
Still not convinced this is a good option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Steve Murphy
Joshua Colp
Russell Bryant
Luigi Rizzo
Matt O'Gorman
Kevin P. Fleming
Jason Parker
Olle Johansson
Matthew Fredrickson