Age | Commit message (Collapse) | Author | Files | Lines |
|
(closes issue #16479)
Reported by: alexrecarey
(closes SWP-577)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@237697 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(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
|
|
(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
|
|
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
|
|
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
|
|
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: http://svn.digium.com/svn/asterisk/branches/1.4@235421 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@233879 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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: http://svn.digium.com/svn/asterisk/branches/1.4@233782 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
In ast_uri_encode, non 7-bit clean characters were being hex escaped
correctly, but control characters were not.
(issue #16299)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@233609 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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: http://svn.digium.com/svn/asterisk/branches/1.4@233092 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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
|
|
(closes issue #16264)
Reported by: dimas
Patches:
event-ack.patch uploaded by dimas (license 88)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232581 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #16290)
Reported by: wdoekes
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232350 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #16367)
Reported by: falves11
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232165 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@232007 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231926 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
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: http://svn.digium.com/svn/asterisk/branches/1.4@231911 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
non-null data ptr
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231853 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
know formats are found.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231740 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
app_voicemail from entering an infinite loop when the same format is specified twice in the format list.
(closes issue #15625)
Reported by: Shagg63
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/429/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231614 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
AST-2009-010
(closes issue #16242)
Reported by: amorsen
Patches:
issue16242.diff uploaded by oej (license 306)
Tested by: amorsen, oej, dvossel
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231441 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@231298 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Spotted by Stuart Henderson
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@230469 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Solaris will crash if NULL is passed to ast_log. This simple patch simply uses S_OR to
get around this.
(closes issue #15392)
Reported by: yrashk
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@229498 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
characters, they will compare equal.
The example given in the issue report is that of [234] and [246], which have
these characteristics, yet they are clearly not equivalent. The code still
uses these two characteristics, yet when the two scores compare equal, an
additional check will be done to compare all characters within the class to
verify equality.
(closes issue #15421)
Reported by: jsmith
Patches:
20091109__issue15421__2.diff.txt uploaded by tilghman (license 14)
Tested by: jsmith, thedavidfactor
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@229360 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Update a WARNING message to give a suggested fix when encountered.
(closes issue #16198)
Reported by: atis
Tested by: atis
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228896 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
After writing to the audiohook list in ast_write(), frames
were being freed incorrectly. Under certain conditions this
resulted in a double free crash.
(closes issue #16133)
Reported by: wetwired
(closes issue #16045)
Reported by: bluecrow76
Patches:
issue16045.diff uploaded by dvossel (license 671)
Tested by: bluecrow76, dvossel, habile
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228692 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
passing through the Asterisk core.
(closes issue #15560)
Reported by: jvandal
(closes issue #15709)
Reported by: covici
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228409 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
strings returned from BASE64_DECODE.
(closes issue #15271)
Reported by: chappell
Patches:
base64_fix.patch uploaded by chappell (license 8)
Tested by: kobaz
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228378 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #15981)
Reported by: slavon
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228338 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
the filestream.
(closes issue #15495)
Reported by: pdf
Patches:
20090916__issue15495.diff.txt uploaded by tilghman (license 14)
Tested by: pdf
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226138 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225171 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225169 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #16103)
Reported by: majorbloodnok
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225105 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
The reasoning for these changes are the same as what I wrote in the commit
message for rev 222878.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224931 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
While this looks like an optimization, it prevents a crash from occurring
when used with certain audiohook callbacks (diagnosed with SVN trunk,
backported to 1.4 to keep the source consistent across versions).
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224855 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
While testing some endpoints that support 16kHz and 32kHz sample rates, some
log messages were generated due to calc_rxstamp() computing timestamps in a way
that produced odd results, so this patch sanitizes the result of the
computations.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224670 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223486 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
The purpose of this code was to have a hangup frame put on the list of deferred
frames. However, the code that read the hangup frame was outside of the scope
of where the hangup frame was declared.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223485 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #15104)
Reported by: nblasgen
Patches:
manager-timeout1.diff uploaded by mnicholson (license 96)
Tested by: nblasgen, mnicholson
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223225 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
This patch is related to a number of issues on the bug tracker that show
crashes related to freeing frames that came from a filestream. A number of
fixes have been made over time while trying to figure out these problems, but
there re still people seeing the crash. (Note that some of these bug reports
include information about other problems. I am specifically addressing
the filestream frame crash here.)
I'm still not clear on what the exact problem is. However, what is _very_
clear is that we have seen quite a few problems over time related to unexpected
behavior when we try to use embedded frames as an optimization. In some cases,
this optimization doesn't really provide much due to improvements made in other
areas.
In this case, the patch modifies filestream handling such that the embedded frame
will not be returned. ast_frisolate() is used to ensure that we end up with a
completely mallocd frame. In reality, though, we will not actually have to malloc
every time. For filestreams, the frame will almost always be allocated and freed
in the same thread. That means that the thread local frame cache will be used.
So, going this route doesn't hurt.
With this patch in place, some people have reported success in not seeing the
crash anymore.
(SWP-150)
(AST-208)
(ABE-1834)
(issue #15609)
Reported by: aragon
Patches:
filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2)
Tested by: aragon, russell
(closes issue #15817)
Reported by: zerohalo
Tested by: zerohalo
(closes issue #15845)
Reported by: marhbere
Review: https://reviewboard.asterisk.org/r/386/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@222878 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
ABE-1998
Review: https://reviewboard.asterisk.org/r/395/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@222877 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@222152 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
suck.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@221970 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@221776 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #15865)
Reported by: kobaz
Patches:
20090915__issue15865.diff.txt uploaded by tilghman (license 14)
Tested by: kobaz
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@221200 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@221086 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
Call Progress() in your dialplan if you explicitly want progress to be sent.
(Reverts change 216430, closes issue #15957)
Reported by: Pavel Troller on the Asterisk-Dev mailing list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@220288 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
(closes issue #15129)
Reported by: bmh
Patches:
20090918__issue15129.diff.txt uploaded by tilghman (license 14)
Review:
https://reviewboard.asterisk.org/r/372/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@219653 f38db490-d61c-443f-a65b-d21fe96a405b
|
|
removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316)
Reported by: vmarrone
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@219136 f38db490-d61c-443f-a65b-d21fe96a405b
|