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r170648 | file | 2009-01-23 16:16:39 -0400 (Fri, 23 Jan 2009) | 4 lines
When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them.
(closes issue #14249)
Reported by: RadicAlish
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r170392 | mmichelson | 2009-01-23 09:40:39 -0600 (Fri, 23 Jan 2009) | 28 lines
Fix broken call pickup
There was a subtle change in ast_do_masquerade which
resulted in failed attempts to pickup calls. The problem
was that the value of the AST_FLAG_OUTGOING flag was
copied from the clone to the original channel. In the case
of call pickup, this meant that the AST_FLAG_OUTGOING flag
ended up being cleared on the channel that was attempting
to execute the pickup.
Because this flag was not set, when ast_read came across
an answer frame, it ignored it. The result of this was that
the calling channel was never properly answered.
This fix changes the behavior in ast_do_masquerade to set
the flags on the original channel to the union of the flags
on the clone channel. This way, if the AST_FLAG_OUTGOING
flag is set on either of the two channels involved in the
masquerade, the resulting channel will have the flag set
as well.
(closes issue #14206)
Reported by: francesco_r
Patches:
14206.patch uploaded by putnopvut (license 60)
Tested by: francesco_r, aragon, putnopvut
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r166568 | mmichelson | 2008-12-23 09:16:26 -0600 (Tue, 23 Dec 2008) | 12 lines
Fix a crash resulting from a datastore with inheritance but no duplicate callback
The fix for this is to simply set the newly created datastore's data pointer
to NULL if it is inherited but has no duplicate callback.
(closes issue #14113)
Reported by: francesco_r
Patches:
14113.patch uploaded by putnopvut (license 60)
Tested by: francesco_r
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This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor
continue recording the call even after the transfer
has completed.
It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.
(closes issue #13538)
Reported by: mbit
Patches:
13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/102/
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r164201 | russell | 2008-12-15 08:31:37 -0600 (Mon, 15 Dec 2008) | 31 lines
Handle a case where a call can be bridged to a channel that is still ringing.
The issue that was reported was about a case where a RINGING channel got
redirected to an extension to pick up a call from parking. Once the parked
call got taken out of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing indication. This patch
fixes this case so that the caller receives ringback once it comes out of
parking until the other side answers.
The fixes are:
- Make sure we remember that a channel was an outgoing channel when doing
a masquerade. This prevents an erroneous ast_answer() call on the channel,
which causes a bogus 200 OK to be sent in the case of SIP.
- Add some additional comments to explain related parts of code.
- Update the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that are events
or commands should be ignored.
- When a bridge first starts, check to see if the peer channel needs to be
given ringing indication because the calling side is still ringing.
- Rework ast_indicate_data() a bit for the sake of readability.
(closes issue #13747)
Reported by: davidw
Tested by: russell
Review: http://reviewboard.digium.com/r/90/
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r163448 | russell | 2008-12-12 07:44:08 -0600 (Fri, 12 Dec 2008) | 26 lines
Resolve issues that could cause DTMF to be processed out of order.
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
(closes issue #12658)
Reported by: dimas
Tested by: russell, file
Review: http://reviewboard.digium.com/r/85/
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The conversion to use ast_check_hangup() everywhere instead of checking the softhangup
flag directly introduced this problem. The issue is that ast_check_hangup() checked
for tech_pvt to be NULL. Unfortunately, this will be NULL is some valid circumstances,
such as with a dummy channel.
The fix is simple. Don't check tech_pvt. It's pointless, because the code path that
sets this to NULL is when the channel hangup callback gets called. This happens inside
of ast_hangup(), which is the same function responsible for freeing the channel. Any
code calling ast_check_hangup() better not be calling it after that point, and if so,
we have a bigger problem at hand.
(closes issue #14035)
Reported by: erogoza
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ast_channel_search_locked need to be made. Specifically, the caller needs to be able to
pass arbitrary data which in turn is passed to the callback. This patch addresses all
of the nested functions currently in asterisk trunk.
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channel list calling a caller-defined callback. The callback returns non-zero
if a match is found. This should speed up some of the code that I committed
earlier today in chan_sip (which is also updated by this commit).
Reviewed by russellb and kpfleming via ReviewBoard:
http://reviewboard.digium.com/r/28/
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or when calling ast_waitfor(). These are inappropriate times to hold the channel
lock. This is what has caused "could not get the channel lock" messages from
chan_sip and has likely caused a negative impact on performance results of SIP
in Asterisk 1.6. Thanks to file for pointing out this section of code.
(closes issue #13287)
(closes issue #13115)
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r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line
After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints.
Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations,
where you'd want to post single-channel cdrs.
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r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines
(closes issue #13409)
Reported by: tomaso
Patches:
asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564)
I basically spent the day, verifying that this patch
solves the problem, and doesn't hurt in non-problem
cases. Why valgrind did not plainly reveal this leak
absolutely mystifies and stuns me.
Many, many thanks to tomaso for finding and providing the fix.
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would reset to 500 ms every time a non-voice frame
was received. The total time we poll should be 500 ms, so
now we save the amount of time left after the poll returned
and use that as our argument for the next call to poll
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Instead, poll the channel until receiving a voice frame. The
cap on this poll is 500 ms.
The optional delay is still allowable in the Answer() application,
but the delay has been moved back to its original position, after
the call to the channel's answer callback. The poll for the voice
frame will not happen if a delay is specified when calling Answer().
(closes issue #12708)
Reported by: kactus
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r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines
Merging the issue11259 branch.
The purpose of this branch was to take into account
"burps" which could cause jitterbuffers to misbehave.
One such example is if the L option to Dial() were used
to inject audio into a bridged conversation at regular
intervals. Since the audio here was not passed through
the jitterbuffer, it would cause a gap in the jitterbuffer's
timestamps which would cause a frames to be dropped for a
brief period.
Now ast_generic_bridge will empty and reset the jitterbuffer
each time it is called. This causes injected audio to be handled
properly.
ast_generic_bridge also will empty and reset the jitterbuffer
if it receives an AST_CONTROL_SRCUPDATE frame since the change
in audio source could negatively affect the jitterbuffer.
All of this was made possible by adding a new public API call
to the abstract_jb called ast_jb_empty_and_reset.
(closes issue #11259)
Reported by: plack
Tested by: putnopvut
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r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines
Revert inadvertent changes to app_skel that occurred when
I was testing for a memory leak
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r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines
Remove properties that should not be here
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r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines
(closes issue #12982)
Reported by: bcnit
Tested by: murf
I discovered that also, in the previous bug fixes and changes,
the cdr.conf 'unanswered' option is not being obeyed, so
I fixed this.
And, yes, there are two 'answer' times involved in this
scenario, and I would agree with you, that the first
answer time is the time that should appear in the CDR.
(the second 'answer' time is the time that the bridge
was begun).
I made the necessary adjustments, recording the first
answer time into the peer cdr, and then using that to
override the bridge cdr's value.
To get the 'unanswered' CDRs to appear, I purposely
output them, using the dial cmd to mark them as
DIALED (with a new flag), and outputting them if
they bear that flag, and you are in the right mode.
I also corrected one small mention of the Zap device
to equally consider the dahdi device.
I heavily tested 10-sec-wait macros in dial, and
without the macro call; I tested hangups while the
macro was running vs. letting the macro complete
and the bridge form. Looks OK. Removed all the
instrumentation and debug.
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r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines
Fix some errant device states by making the devicestate API more strict in
terms of the device argument (only without the unique identifier appended).
(closes issue #12771)
Reported by: davidw
Patches:
20080717__bug12771.diff.txt uploaded by Corydon76 (license 14)
Tested by: davidw, jvandal, murf
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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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callback. This change differs from commit 127113 in that now the
channel is not set to AST_STATE_UP until after the answer callback.
(closes issue #12924)
Reported by: snyfer
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- change ast_settimeout() to honor max rate in edge cases of file playback
(this will make some warning messages go away at the end of playing back
a file)
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r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines
allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places
don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it
get app_rpt building again after the DAHDI changes
(closes issue #12911)
Reported by: tzafrir
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- Convert the last part of channel.c over to use the timing API. This would
not have made a difference when using the dahdi timing module. I noticed
it when trying to use another timing source. Oops. :)
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- Convert chan_iax2 to use the timing API
- Convert usage of timing in the core to use the timing API instead of
using DAHDI directly
- Make a change to the timing API to add the set_rate() function
- change the timing core to use a rwlock
- merge a timing implementation, res_timing_dahdi
Basic testing was successful using res_timing_dahdi
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r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines
Occasionally, the alertpipe loses its nonblocking status, so detect and correct
that situation before it causes a deadlock. (Reported and tested by ctooley
via #asterisk-dev)
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r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines
Do not attempt to do emulation if an END digit is received and the length is
less than the defined minimum digit length, and the other end only wants END
digits (SIP INFO, for example).
(closes issue #12778)
Reported by: tsearle
Patches:
12778.rev1.txt uploaded by russell (license 2)
Tested by: tsearle
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hold tracking information for mutexes. Now, the "core show locks" output
will output information about who is holding a rwlock when a thread is
waiting on it.
(closes issue #11279)
Reported by: ys
Patches:
trunk_lock_utils.v8.diff uploaded by ys (license 281)
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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