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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85552 | file | 2007-10-15 11:55:04 -0300 (Mon, 15 Oct 2007) | 4 lines
If Monitor or a spy was added to a P2P or native bridged channel bring the channel back to the generic bridging core so the monitor or spy operations work.
(closes issue #10943)
Reported by: julianjm
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Closes issue #10913, reported by tootai, who graciously granted us access
to his Asterisk server, thanks! Daniel, feel free to reopen the bug in
case you can reproduce this on 1.4.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Many uses of the astlisting environment around verbatim text to ensure that
it gets properly formatted and doesn't run off the page.
- Update some things that have been deprecated.
- Add escaping as needed
- and more ...
(closes issue #10978)
Reported by: IgorG
Patches:
texdoc-85542-1.patch uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) | 12 lines
Fix an issue with console verbosity when running asterisk -rx to execute a command
and retrieve its output. The issue was that there was no way for the main Asterisk
process to know that the remote console was connecting in the -rx mode. The way that
James has fixed this is to have all remote consoles muted by default. Then, regular
remote consoles automatically execute a CLI command to unmute themselves when they
first start up.
(closes issue #10847)
Reported by: atis
Patches:
asterisk-consolemute.diff.txt uploaded by jamesgolovich (license 176)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Added development section (backtrace.tex)
- Correct filesystem path formating
- Replace all "|" argument separator to ","
- Endless count of spaces at the end of line
- Using astlisting to make listings do not take so much place
- Take back ASTRISKVERSION on first page
- Make localchannel.tex readable by inserting extra end of lines
(closes issue #10962)
Reported by: IgorG
Patches:
texdoc-85177-1.patch uploaded by IgorG (license 20)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
