Age | Commit message (Collapse) | Author |
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r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, 06 Apr 2012) | 2 lines
Fix typo in svn:keywords
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r361412 | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 lines
Fix typo in svn:keywords
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Added a '\n' to the warning messages when we ignore a media stream due to the
port number being '0'.
(closes issue ASTERISK-19646)
Reported by: Badalian Vyacheslav
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The error message for failure to stop autoservice after a gosub or macro call
during a dial was removed for macro while Asterisk 1.4 was still being actively
developed. The corresponding gosub error message was never removed.
(closes issue ASTERISK-19551)
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There were a few instances of restarting music on hold in meetme that would cause
Asterisk to revert to the default class of music on hold for no adequate reason.
Review: https://reviewboard.asterisk.org/r/1844/
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The important parts of the patch were already applied through other updates.
(closes issue ASTERISK-19445)
Reported by: Makoto Dei
Patches:
memset-memcpy-length.patch uploaded by Makoto Dei (license 5027)
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(closes issue ASTERISK-19444)
Reported by: Makoto Dei
Patches:
devstate-change-usage-truncate.patch uploaded by Makoto Dei (license 5027)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue ASTERISK-19540)
Reported by: Makoto Dei
Patches:
clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
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Also add from the patch the portion in res_fax_spandsp that didn't apply to 1.8
Merged revisions 361142 from http://svn.asterisk.org/svn/asterisk/branches/1.8
(closes issue ASTERISK-19540)
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(Closes Issue ASTERISK-19335)
Reported by: Johan Wilfer
Review: https://reviewboard.asterisk.org/r/1843/
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This came up while fixing documentation generation for many other cases where
the argument separator was not being displayed properly. Now that it is
displayed properly, it shows up in the wrong place for Transfer since the '/'
is only required if Tech is present.
(related to issue ASTERISK-18168)
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(related to ASTERISK-19575)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Hangup now can take a regular expression as the Channel option. If you want
to hangup multiple channels, use /regex/ as the Channel option. Existing
behavior to hanging up a single channel is unchanged, but if you pass a regex,
the manager will send you a list of channels back that were hung up.
(closes issue ASTERISK-19575)
Reported by: Mark Murawski
Tested by: Mark Murawski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This change prevents Asterisk from sending RTCP receiver reports during a
remote bridge since it is no longer receiving media and should not be
reporting anything.
(related to ASTERISK-19366)
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The logger_thread() had an exit path that failed to release the logmsgs
list lock.
* Make logger_thread() exit path unlock the logmsgs list lock.
* Made ast_log() not queue any messages to the logmsgs list if the
close_logger_thread flag is set.
(issue ASTERISK-19463)
Reported by: Matt Jordan
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Prior to this patch, a connected line update was queued during
call pickup and then an answer frame was queued. The original
caller would presumably then have his connected line updated
and then the call would be answered.
In actuality, the answer frame was not how the call ended up
being answered. Rather, an odd section in app_dial that checks
if the called channel's state is up.
The result is that the order of the connected line update and
the answer were variable. In most cases, this wasn't actually
a bad thing. However, if the 'I' option was passed to dial, the
connected line update would be inhibited.
The fix is to queued the connected line after the answer frame is
queued. This way the race in app_dial is between two
conditions resulting in an answer. This way the connected line
update occurs after the answer every time.
(closes issue ASTERISK-19183)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Mark Michelson
Patches:
ASTERISK-19183.patch uploaded by Mark Michelson (license 5049)
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requests.
This change makes use of connected party information in addition to caller ID in order
to populate local and remote XML elements in the dialog-info NOTIFYs.
(closes issue ASTERISK-16735)
Reported by: Maciej Krajewski
Tested by: Maciej Krajewski
Patches:
local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
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* Rename astobj2 API parameter funcname to func.
* Rename astobj2 API iterator parameter to iter.
* Update some documentation for OBJ_MULTIPLE.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Log messages will now display a call number that they are tied to (ordered for calls
based on when they started). This feature is made to be minimally invasive without
requiring changes to many of the existing log messages. These IDs won't show up for
verbose messages on CLI (but they will in log files) This is currently in phase II
of production, see more about this feature on the wiki --
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
Review: https://reviewboard.asterisk.org/r/1823/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Log messages will now display a call number that they are tied to (ordered for calls
based on when they started). This feature is made to be minimally invasive without
requiring changes to many of the existing log messages. These IDs won't show up for
verbose messages on CLI (but they will in log files) This is currently in phase II
of production, see more about this feature on the wiki --
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
Review: https://reviewboard.asterisk.org/r/1823/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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A previous CDR fix for setting CDR variables during a bridge via
custom dialplan features broke setting CDR variables in the
hangup extension. This patch fixes the issue.
Review: https://reviewboard.asterisk.org/r/1794/
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I was getting confused during some testing why Asterisk was saying that
a subscription was being added when it was clearly being removed. This
fixes that confusion.
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Global ao2 objects must always exist after initialization because there is
no access control to obtain another reference to the global object.
It is expected that module configuration could use these new API calls to
replace an active configuration parameter object with an updated
configuration parameter object.
With these new API calls, the global object could be replaced, removed, or
referenced without the risk of someone using a stale global object
pointer.
Review: https://reviewboard.asterisk.org/r/1824/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(issue ASTERISK-17842)
Reported by: Bryon Clark
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Update CHANGES for r360471
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Rather then flood the CLI with verbose messages, we've changed the level to
debug. This will help keep the CLI clean.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes ASTERISK-17842)
Reported by: Bryon Clark
Patches:
20110512__issue19278.diff.txt uploaded by Tilghman Lesher (license 5003)
configure_bfd_with_dl_and_iberty.patch uploaded by Bryon Clark (license 6157)
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This is just a minor code cleanup change. These uses of ao2_callback() would
never return anything since the callbacks always returned 0. However, be more
explicit that no returned results are wanted by specifying OBJ_NODATA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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dial_list is a dynamically allocated array that is allocated at the beginning
of Page() based on how many devices will be dialed. This was never being freed.
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r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) | 6 lines
expression parser: Fix (theoretical) memory leak.
Fix a memory leak that is very unlikely to actually happen. If a malloc()
succeeded, but the following strdup() failed, the memory from the original
malloc() would be leaked.
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r360357 | russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
Rebuild parsers.
This is needed to include the last fix to main/ast_expr2.y. The changes look
much bigger as this regeneration of the code was done with newer versions of
flex and bison.
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Q.951 indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening indicator
field should be "Network provided".
* Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to interworking". This
fix makes Asterisk consistent and it also makes it consistent with earlier
branches as far as this presentation value is concerned.
* Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions
handle the "Number not available due to interworking" case better in
sig_pri.c. This change is possible because the minimum required libpri
version (v1.4.11) has the necessary defines in libpri.h.
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Similar to dial and queue F option.
(Closes issue ASTERISK-19282)
Reported by: To
Patches:
bridge_f-v3.diff uploaded by To (license 6347)
Review: https://reviewboard.asterisk.org/r/1825/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Everything still compiled after making these changes, so I assume these
whitespace-only changes didn't break anything (and shouldn't have).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@360190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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move SQLite3.
(closes issue ASTERISK-19367)
Reported by: Andrew Latham
Patches:
debian_install_prereq.diff uploaded by Andrew Latham (license 5985)
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Also detect gmime version 2.6 (Michael Biebl)
Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>
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When Asterisk detects a hangup and cannot send a BYE due to a pending
INVITE, it sets the pendingbye flag and waits for the final response to that
INVITE. When the response is received, it transmits the BYE. If, however,
that INVITE request is a pending re-INVITE, it needs to first send a CANCEL
request to terminate the pending re-INVITE. In that circumstance, Asterisk
was, in some scenarios, clearing the pendingbye flag after processing the
CANCEL request and not checking for a pending BYE when receiving the final
487 response to the INVITE.
This patch ensures that if the pendingbye flag is set, it is honored
regardless of the nature of the INVITE request currently in flight.
(closes issue ASTERISK-19365)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283)
Review: https://reviewboard.asterisk.org/r/1807
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Echo()'s description states that it echoes audio, video, and DTMF except for #
while it actually echoes any frame that it receives other than DTMF #. This
was causing frame storms in the test suite in some circumstances where Echo()
was attached to both ends of a pair of local channels and control frames
were being periodically generated. Echo()'s behavior and description have
been modifed so that it only echoes media and non-# DTMF frames.
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The PeerStatus event for IAX2 channels currently includes a header named Post
which should have been Port. Post was removed and the AMI version has been
updated to 1.3.
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Fix AMI module reload deadlock regression from ASTERISK-18479 when it
tried to fix the race between calling an AMI action callback and
unregistering that action. Refixes ASTERISK-13784 broken by
ASTERISK-17785 change.
Locking the ao2 object guaranteed that there were no active callbacks that
mattered when ast_manager_unregister() was called. Unfortunately, this
causes the deadlock situation. The patch stops locking the ao2 object to
allow multiple threads to invoke the callback re-entrantly. There is no
way to guarantee a module unload will not crash because of an active
callback. The code attempts to minimize the chance with the registered
flag and the maximum 5 second delay before ast_manager_unregister()
returns.
The trunk version of the patch changes the API to fix the race condition
correctly to prevent the module code from unloading from memory while an
action callback is active.
* Don't hold the lock while calling the AMI action callback.
(closes issue ASTERISK-19487)
Reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1818/
Review: https://reviewboard.asterisk.org/r/1820/
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* Added missing error exits with cause in manager_mutestream().
* Cleaned up manager_mutestream() and func_mute_write().
* Some whitespace and comment cleanup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch addresses a bug with chanspy on local channels which roughly 50% of the time
would create a situation where chanspy can latch onto a zombie channel, keeping the zombie
alive forever and causing the channel doing the spying to never be able to hang up.
(closes issue ASTERISK-19493)
Reported by: lvl
Review: https://reviewboard.asterisk.org/r/1819/
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* Remove unnnecessary const from const char * const var declaration in the
ast_app_run_macro() and ast_app_run_sub() prototypes. The second const is
unnecessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The code may be just fine, but it had not received a "ship it!" on
review board yet.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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from Review: https://reviewboard.asterisk.org/r/1699/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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There exists a remotely exploitable stack buffer overflow in HTTP digest
authentication handling in Asterisk. The particular method in question
is only utilized by HTTP AMI. When parsing the digest information, the
length of the string is not checked when it is copied into temporary buffers
allocated on the stack.
This patch fixes this behavior by parsing out pre-defined key/value pairs
and avoiding unnecessary copies to the stack.
(closes issue ASTERISK-19542)
Reported by: Russell Bryant
Tested by: Matt Jordan
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