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TLS needs to have one configuration per configured domain at some point.
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Actually, kill the in-memory structure for type=user and start using the sip_peer
structure for every object. Have only one in-memory list and use them different
ways depending on type=user, type=peer and type=friend - like always.
The idea with this first patch is that configurations should work as before.
Some additional features for realtime peers. By adding a type= field, you
can now have multiple functions.
Let's test this for a while. Won't be integrated in 1.6.0, only in trunk,
for testing.
There's propably more to clean up and simplify here. Help is welcome
and encouraged!
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AMI commands can display that a channel is under control of an AGI.
Work inspired by work at customer site, but paid for by Edvina AB
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(if we have RTCP reports
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.
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(closes issue #12860)
Reported by: flyn
Patches:
asterisk.ldif uploaded by suretec (license 70)
asterisk.schema uploaded by suretec (license 70)
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file...
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application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset
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r128029 | tilghman | 2008-07-04 11:11:03 -0500 (Fri, 04 Jul 2008) | 2 lines
Move the free down one
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) | 8 lines
Fix the 'dialplan remove extension' logic, so that it a) works with cidmatch,
and b) completes contexts correctly when the extension is ambiguous.
(closes issue #12980)
Reported by: licedey
Patches:
20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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the IANA list
- add a few new registered and well-known extension names
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul 2008) | 6 lines
a couple of small Solaris-related fixes
(closes issue #11885)
Reported by: snuffy, asgaroth
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r127895 | kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 lines
remove this, it has been moved to the main Makefile
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of the whisper or barge audiohooks fails.
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(closes issue #12986)
Reported by: andrew53
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines
The CDRfix4/5/6 omnibus cdr fixes.
(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror
(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11
(closes issue #11849)
Reported by: greyvoip
As to 11849, I think these changes fix the core problems
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.
Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.
(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf
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May fix issue 12919.
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should not have session-timer
require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses
that are related to INVITEs and re-INVITEs.
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1.4 since the equivalent is already here)
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explanation of the change may be found in configs/queues.conf.sample
(closes issue #12690)
Reported by: atis
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It's mixing peers and users in a strange way and should really not be a CLI command,
since it's not meant for human output. It should be done with an app connecting to
manager.
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sizeof() and division.
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(Closes issue #12892)
Reported by: jaroth
Patch originally by jaroth, fixed by me.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul 2008) | 3 lines
Fix thread-safety of some of the pbx_builtin_getvar_helper calls
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acceptdtmf and enddtmf. These allow for the DTMF pressed to be configurable
instead of being hardcoded to '#' and '*'.
(AST-86)
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http support
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r127501 | tilghman | 2008-07-02 14:47:15 -0500 (Wed, 02 Jul 2008) | 14 lines
Merged revisions 127466 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r127466 | tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
Solaris fix
(closes issue #12949)
Reported by: snuffy
Patches:
bug_12949.diff uploaded by snuffy (license 35)
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(closes issue #12949)
Reported by: snuffy
Patches:
bug_12949.diff uploaded by snuffy (license 35)
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to document arguments seems logical, and does work, but is not the best
thing to use.
\arg in doxygen is simply for creating non-nested unordered lists. \param is
the correct tag to use to document function parameters, and will come out
better in the generated documentation.
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This section covers some locking fundamentals, as well as some information on
locking as it is used in Asterisk. It describes some of the ways that are used
and could be used to achieve deadlock avoidance. It also demonstrates the
unfortunate conclusion that with the use of recursive locks, none of the
constructs in use today are failsafe from deadlocks. Finally, it makes some
recommendations for new code being written. As proper locking strategies is a
complex subject, this section still has room for expansion and improvement.
This is a result of collaboration between Luigi Rizzo and myself on the
asterisk-dev mailing list.
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We can still send video, just don't try to use what is not available.
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specific configuration in oss.conf
(reported by Klaus Darillion on the -video mailing list).
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recommended in RFC 3261, instead of being hardcoded to 32 seconds. This is
important for LANs, as it allows autocongestion to occur much more quickly, if
desired by the local PBX administrator. It also corrects a bug: if the T1
timer was increased beyond 500ms, then timer B would have been set at a much
lower value than recommended.
(closes issue #12544)
Reported by: kactus
Patches:
20080616__bug12544.diff.txt uploaded by Corydon76 (license 14)
Tested by: kactus
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