https://origsvn.digium.com/svn/asterisk/trunk
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r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@252175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, we only created the default /var/run/asterisk directory at install
time. While we could create it in the init script, that would not work for
those who start asterisk manually from the command line. So the safest thing
to do is to create it as part of the Asterisk boot process. This also changes
the ownership of the directory, because the pid and ctl files are created after
we setuid/setgid.
(closes issue #16802)
Reported by: Brian
Patches:
20100224__issue16802.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@248859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On channel destruction the channel's datastores are removed and
destroyed. Since there are public API calls to find and remove
datastores on a channel, a lock should be held whenever datastores are
removed and destroyed. This resolves a crash caused by a race
condition in app_chanspy.c.
(closes issue #16678)
Reported by: tim_ringenbach
Patches:
datastore_destroy_race.diff uploaded by tim ringenbach (license 540)
Tested by: dvossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@246545 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Branch support, retains ABI, if backend CDR collector is adaptive then database
requires 'dnid' field to be added, otherwise no functional changes.
Reported by: alecdavis
Tested by: alecdavis
Patch
cdr_dnid.diff2.txt uploaded by alecdavis (license 585)
Review: https://reviewboard.asterisk.org/r/455/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@242142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call,
by executing ast_cdr_update() early in __ast_pbx_run.
Based on cdr_update.diff3.txt
(issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@241458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
While reading through configuration files with the intent of returning their
full contents (comments specifically) we allocated some memory and then forgot
to free it. This doesn't fix 16554 but clears up a leak I had in the lab.
(issue #16554)
Reported by: mav3rick
Patches:
issue16554_20100118.patch uploaded by seanbright (license 71)
Tested by: seanbright
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@241015 65c4cc65-6c06-0410-ace0-fbb531ad65f3
asterisk.conf's 'transmit_silence' option existed before
this patch, but was limited to only generating silence
while recording and sending DTMF. Now enabling the
transmit_silence option generates silence during wait
times as well.
To achieve this, ast_safe_sleep has been modified to
generate silence anytime no other generators are present
and transmit_silence is enabled. Wait apps not using
ast_safe_sleep now generate silence when transmit_silence
is enabled as well.
(closes issue 0016524)
Reported by: kobaz
(closes issue 0016523)
Reported by: kobaz
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/456/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@239718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is in a section of code that relates to two other issues, namely
issue #14011 and issue #14940), one of which was the behavior of
Background when called with a context argument that matched the current
context. This fix broke FreePBX, however, in a post-Dial situation.
Needless to say, this is an extremely difficult collision of several
different issues. While the use of an exception flag is ugly, fixing all
of the issues linked is rather difficult (although if someone would like
to propose a better solution, we're happy to entertain that suggestion).
(closes issue #16434)
Reported by: rickead2000
Patches:
20091217__issue16434.diff.txt uploaded by tilghman (license 14)
20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: rickead2000
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@237405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch is simple in that it reorders the disposition defines so that the fix
for issue 12946 works properly (the default CDR disposition was changed to
AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to
ensure all CDR records are written.
The side effects of CDR changes are scary, so I'm documenting the test cases
performed to attempt to catch any regressions. The following tests were all
performed using 1.4 rev 195881 vs head (235571) + patch:
A calls B
C calls B (busy)
Hangup C
Hangup A
(Both SIP and features)
A calls B
A blind transfers to C
Hangup C
(Both SIP and features)
A calls B
A attended transfers to C
Hangup C
A calls B
A attended transfers to C (SIP)
C blind transfers to A (features)
Hangup A
All of the test scenario CDRs matched.
The following tests were performed just with the patch to ensure proper operation
(with unanswered=yes):
exten =>s,1,Answer
exten =>s,n,ResetCDR(w)
exten =>s,n,ResetCDR(w)
exten =>s,1,ResetCDR(w)
exten =>s,n,ResetCDR(w)
(closes issue #16180)
Reported by: aatef
Patches:
bug16180.patch uploaded by jpeeler (license 325)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@235635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also, ensure that the extension COULD match, not just that it won't match more.
(closes issue #16113)
Reported by: OrNix
Patches:
20091216__issue16113.diff.txt uploaded by tilghman (license 14)
Tested by: OrNix
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@235421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent change to app_voicemail made it such that the module now assumes that
all format modules are available while processing voicemail configuration.
However, when autoloading modules, it was possible that app_voicemail was
loaded before the format modules. Since format modules don't depend on
anything, set a module load priority on them to ensure that they get loaded
first when autoloading.
This version of the patch is specific to Asterisk 1.4 and 1.6.0. These versions
did not already support module load priority in the module API. This adds a
trivial version of this which is just a module flag to include it in a pass before
loading "everything".
Thanks to mmichelson for the review!
(closes issue #16412)
Reported by: jiddings
Tested by: russell
Review: https://reviewboard.asterisk.org/r/445/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@233782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This code was added for helping to debug the source of invalid HOLD frames.
However, a side effect of this is that it will incorrectly report errors for
frames that have an integer payload. Make the check for this block specific
to the HOLD frame case.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@233092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added additional field 'int display_inband_dtmf_warning', which when set to '1' displays the warning ('Inband DTMF is not supported on codec %s. Use RFC2833'), and when set to '0' doesn't display the warning. Otherwise you would get hundreds of warnings every second.
(closes issue #15769)
Reported by: falves11
Patches:
patch_15769_14.txt uploaded by mnick (license 874)
Tested by: mnick, falves11
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@233014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The crash was happening as a result of a frame containing an invalid data
pointer, but was set with data length of zero. The few times the issue was
reproduced it _seemed_ that the frame was queued properly, that is the data
pointer was set to NULL. I never could reproduce the crash so as a last resort
the crash has been fixed, but a check in __ast_read has been added to give as
much information about the source of problematic frames in the future.
(closes issue #16058)
Reported by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@231911 65c4cc65-6c06-0410-ace0-fbb531ad65f3
