https://origsvn.digium.com/svn/asterisk/branches/1.4
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r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | 8 lines
If Asterisk is in the middle of shutting down, respond to OPTIONS
with 503 Unavailable.
(closes issue #10994)
Reported by: eserra
Patches:
sip-options-503.patch uploaded by eserra (license 45)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r86028 | mmichelson | 2007-10-16 17:49:10 -0500 (Tue, 16 Oct 2007) | 6 lines
Adding deprecated warning to monitor-join option, since the plan is to
no longer support this in favor of monitor-type = mixmonitor
(related to issue #10885)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85994 | russell | 2007-10-16 17:14:36 -0500 (Tue, 16 Oct 2007) | 16 lines
Some locking errors exposed the fact that the lock debugging code itself was
not thread safe. How ironic! Anyway, these changes ensure that the code that
is accessing the lock debugging data is thread-safe.
Many thanks to Ivan for finding and fixing the core issue here, and also
thanks to those that tested the patch and provided test results.
(closes issue #10571)
(closes issue #10886)
(closes issue #10875)
(might close some others, as well ...)
Patches: (from issue #10571)
ivan_ast_1_4_12_rel_patch_lock.h.diff uploaded by Ivan (license 229)
- a few small changes by me
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85958 | mmichelson | 2007-10-16 16:14:34 -0500 (Tue, 16 Oct 2007) | 5 lines
Trying to remove a non-dynamic queue member via dynamic means can lead to some
interesting (read nasty) situations. This patch clears up the issue by making
only dynamic queue members removable via dynamic methods.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Don't improperly memset() over an ast_str. This was leftover from before it
got changed to use ast_str.
(closes issue #11003, reported by pj)
(closes issue #10770, reported by yehavi)
(patched by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Modifications :
- set bridge type to partial ;
- process media candidates from the remote peer properly.
Now we have Jingle audio, at least between two Asterisk Jingle
clients.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85777 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r85687 | tilghman | 2007-10-15 15:29:35 -0500 (Mon, 15 Oct 2007) | 5 lines
Don't execute a gosub if the arguments is zero-len (not just NULL)
Reported by davevg
Fixed by me
Closes issue #10985
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r85717 | russell | 2007-10-15 15:59:27 -0500 (Mon, 15 Oct 2007) | 7 lines
Previously, app_queue created a thread to handle every single device state
change. I changed this a while ago in trunk for performance reasons. However,
bug 8407 points out that it is actually a race condition, causing device state
changes to get processed in random order. So, I backported my changes from
trunk to 1.4.
(closes issue #8407, patch provided by tim_ringenbach, committed patch by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
It allows you to configure a prefix for auto-monitor recordings.
(closes issue #6353)
Reported by: ivanfm
Patches:
asterisk_automon_v4.patch uploaded by ivanfm (original patch)
- updated patch:
6353-touch_monitor_prefix.diff uploaded by qwell (license 4)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85682 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85647 | russell | 2007-10-15 14:11:38 -0500 (Mon, 15 Oct 2007) | 5 lines
The loop in the handler for the "core show locks" could potentially block for
some amount of time. Be a little bit more careful and prepare all of the
output in an intermediary buffer while holding a global resource. Then, after
releasing it, send the output to ast_cli().
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | 6 lines
Make the default for the srvlookup option to be yes. It doesn't really make
sense for it to default to off. The default configuration file has it on, and
proper RFC behavior, as indicated by a comment in the code, is for it to be on.
So, let's have it on by default to make lives easier.
(closes issue #10954, suggested by jtodd)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85556 | russell | 2007-10-15 10:40:45 -0500 (Mon, 15 Oct 2007) | 9 lines
Ensure the buffer passed to ast_canmatch_extension() is properly initialized so
that it is null terminated.
(issue #10977)
Reported by: dimas
Patches:
pbxdundi.patch uploaded by dimas (license 88)
- small mods by me
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
