https://origsvn.digium.com/svn/asterisk/branches/1.4
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r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) | 1 line
another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users.
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https://origsvn.digium.com/svn/asterisk/trunk
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Allow Spiraled INVITEs to work correctly within Asterisk.
Prior to this change, a spiraled INVITE would cause a 482
Loop Detected to be sent to the caller. With this change,
if a potential loop is detected, the Request-URI is inspected
to see if it has changed from what was originally received. If
pedantic mode is on, then this inspection is fully RFC 3261
compliant. If pedantic mode is not on, then a string comparison
is used to test the equality of the two R-URIs.
This has been tested by using OpenSER to rewrite the R-URI
and send the INVITE back to Asterisk.
(closes issue #7403)
Reported by: stephen_dredge
Modified:
branches/1.4/channels/chan_sip.c
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r132787 | kpfleming | 2008-07-22 17:11:01 -0500 (Tue, 22 Jul 2008) | 5 lines
fix up namespace pollution for dahdi_chan_mode enum
correct registration of AMI actions in chan_dahdi; in zap-only mode, only register the Zap flavors of the actions (and use Zap prefixes for headers and acks), but in dahdi+zap mode, register both Zap and DAHDI flavors of actions
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul 2008) | 6 lines
ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense'
(closes issue #12160)
Reported by: tzafrir
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines
The most common question on the #asterisk iRC channel and on mailing lists
seems to be in regards to an error message when retransmit fails. This
is frequently misunderstood as a failure of Asterisk, not a failure of
the network to reach the other party.
This document tries to assist the Asterisk user in sorting out these
issues by explaining the logic and pointing at some possible
causes. Hopefully, we will get other questions now :-)
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commands that conflicted with adding new features to the newer debug
commaands.
(closes issue #13103)
Reported by: mvanbaak
Patches:
2008071901__issue13103_iax2_set_debug_peer.diff uploaded by
mvanbaak (license 7)
Tested by: bbryant, mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
own repository, and must be installed like any other library for Asterisk to
use. The two modules that require it are codec_resample and app_jack.
To install libresample:
$ svn co http://svn.digium.com/svn/libresample/trunk libresample
$ cd libresample
$ ./configure
$ make
$ sudo make install
This code is currently in our own repository because the build system did not
include the appropriate targets for building a dynamic library or for installing
the library.
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removed early (before the routine to load the configuration was
finished) because a variable wasn't initialized.
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probably not a good idea, as we might run out of stack space. Therefore,
changing this over to use the ast_str infrastructure for buffers is
probably a good idea.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132206 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the same keyword as the other files (patch by eliel).
(closes issue #13104)
Reported by: eliel
Patches:
revision.patch uploaded by eliel (license 64)
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r132042 | tilghman | 2008-07-18 12:33:42 -0500 (Fri, 18 Jul 2008) | 3 lines
Spinlock within the destroy, to allow a scheduled job to continue, if it's
waiting on the mutex which the destroy thread has.
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buffers=<num of buffers>,<policy>
Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate".
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008) | 7 lines
Revert part of issue #5620 (revision 6965) as it appears that it was in error.
This should fix talk call progress on analog lines.
(closes issue #12178)
Reported by: michael-fig
Patches:
20080717__bug12178.diff.txt uploaded by Corydon76 (license 14)
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a feature without specifying a group or feature to register.
(closes issue #13101)
Reported by: eliel
Patches:
features.c.patch uploaded by eliel (license 64)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
instead. DEVICE_STATE is a state change on one server, and DEVICE_STATE_CHANGE is
the "real" state of that device across all servers sharing state. This would have
only been a problem with distributed device state.
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Reported by: murf
Most of this bug was already fixed by Tilghman before
I opened it; Many thanks to Tilghman for his fix
in svn version 125794. That fix cleared up some of the
fields in the lock_info.
This commit changes the address that is stored for the
lock in the lock_info struct, so that it is the same
as that passed into the locking macros. This makes
searching for a lock_info (as in log_show_lock())
by its lock addr possible. The lock_addr field is
infinitely more useful if it is the same as what
is 'publicly' available outside the lock_info code.
Many thanks to kpfleming, putnopvut, and Russell for their
invaluable insights earlier today.
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