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(closes issue 0017052)
Reported by: dvossel
Tested by: dvossel
(closes issue 0016196)
Reported by: atis
(closes issue 0017052)
Reported by: dvossel
Tested by: dvossel
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Take 2, without ABI breakage this time.
Review: https://reviewboard.asterisk.org/r/588/
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Here is a copy and paste of the details from my request on
reviewboard that dealt with these changes:
Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like:
seqno 1: DTMF 1
seqno 2: DTMF 1
seqno 3: DTMF 1
seqno 4: DTMF 1
seqno 6: DTMF 1 (end)
seqno 5: DTMF 1
seqno 7: DTMF 1 (end)
seqno 8: DTMF 1 (end)
Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too:
seqno 9: DTMF 1
seqno 10: DTMF 1 (end)
seqno 11: DTMF 1 (end)
seqno 13: DTMF 2
seqno 12: DTMF 1 (end)
seqno 14: DTMF 2
seqno 15: DTMF 2 (end)
seqno 16: DTMF 2 (end)
seqno 17: DTMF 2 (end)
In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF.
Fix 2. The second change in place is to fix an issue like the following:
seqno 1: DTMF 1
seqno 2: DTMF 1
seqno 3: DTMF 1 (end) *packet lost*
seqno 4: DTMF 1 (end) *packet lost*
seqno 5: DTMF 1 (end) *packet lost*
seqno 6: DTMF 2
When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list.
Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem
Review: https://reviewboard.asterisk.org/r/558
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DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences.
This can be guaranteed by forcing the ABI to no longer change when these compiler flags are set.
An unfortunate side-effect to this is that there is an ABI change here. However, there is some
mitigation. Existing modules *will* fail to load since they would require functions that no
longer exist.
Review: https://reviewboard.asterisk.org/r/508/
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This re-renames ast_rtp_update_source to ast_rtp_new_source
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Reviewboard: https://reviewboard.asterisk.org/r/551/
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https://origsvn.digium.com/svn/asterisk/trunk
........
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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console... difficult.
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: http://svn.digium.com/svn/asterisk/branches/1.4@248859 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16786)
Reported by: dodo
Patches:
logger2.patch uploaded by dodo (license 989)
Tested by: tilghman
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crash Asterisk.
(closes issue #16470)
Reported by: kjotte
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@248012 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/branches/1.4@246545 f38db490-d61c-443f-a65b-d21fe96a405b
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People don't always build Asterisk from source (distro packages, anybody?).
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expired frames in the queue.
(closes issue #16525)
Reported by: kobaz
Patches:
20100126__issue16525.diff.txt uploaded by tilghman (license 14)
20100129__issue16525__1.6.0.diff.txt uploaded by tilghman (license 14)
Tested by: kobaz, atis
(closes issue #16581)
Reported by: ZX81
(closes issue #16681)
Reported by: alexr1
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pbx_builtin_setvar_helper.
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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: http://svn.digium.com/svn/asterisk/branches/1.4@242142 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/branches/1.4@241458 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/branches/1.4@241015 f38db490-d61c-443f-a65b-d21fe96a405b
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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/
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(closes issue #16241)
Reported by: vnovy
Patches:
manager.c.patch uploaded by vnovy (license 922)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@238915 f38db490-d61c-443f-a65b-d21fe96a405b
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format strings.
(closes issue #16560)
Reported by: goldwein
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(closes issue #16479)
Reported by: alexrecarey
(closes SWP-577)
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(closes issue #16407)
Reported by: qwell
Patches:
20100104__issue16407.diff.txt uploaded by tilghman (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@237573 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16482)
Reported by: wdoekes
Patches:
astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
20091223__issue16482.diff.txt uploaded by tilghman (license 14)
Tested by: wdoekes, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@237493 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/branches/1.4@237405 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/branches/1.4@235635 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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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/
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In ast_uri_encode, non 7-bit clean characters were being hex escaped
correctly, but control characters were not.
(issue #16299)
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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.
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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: http://svn.digium.com/svn/asterisk/branches/1.4@233014 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16264)
Reported by: dimas
Patches:
event-ack.patch uploaded by dimas (license 88)
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(closes issue #16290)
Reported by: wdoekes
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(closes issue #16367)
Reported by: falves11
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