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r242521 | tilghman | 2010-01-24 00:40:31 -0600 (Sun, 24 Jan 2010) | 15 lines
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r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
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r238635 | dvossel | 2010-01-08 13:39:30 -0600 (Fri, 08 Jan 2010) | 22 lines
fixes AUDIOHOOK_INHERIT regression
During the process of removing an audiohook from one channel
and attaching it to another the audiohook's status is updated
to DONE and then back to whatever it was previously. Typically
updating the status after setting it to DONE is not a good idea
because DONE can trigger unrecoverable audiohook destruction
events... because of this a conditional check was added to
audiohook_update_status to explicitly prevent the audiohook
from ever changing after being set to DONE. It was this check
that prevented audiohook inherit from work properly though.
Now ast_audiohook_move_by_source is treated as a special exception,
as the audiohook must be returned to its previous status after
attaching it to the new channel. This is only a safe operation
because the audiohook's lock is held the entire time, otherwise
this could cause trouble.
(closes issue #16522)
Reported by: corruptor
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r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) | 23 lines
Merged revisions 237405 via svnmerge from
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r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines
Add a flag to disable the Background behavior, for AGI users.
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
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r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec 2009) | 14 lines
Merged revisions 236585 via svnmerge from
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r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) | 7 lines
Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces.
There was conditional code (based on build platform) to optioinally wrap
PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions
of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add
a configure-time check for it.
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r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) | 55 lines
Merged revisions 235635 via svnmerge from
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r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines
Correct CDR dispositions for BUSY/FAILED
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)
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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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r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines
Another round of UDPTL stack fixes/improvements:
1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL
session, so that log/error/debug messages generated by the UDPTL stack can
be 'connected' to the endpoint that caused them to be generated.
2) Improve comments (and process) of calculating the far end's maximum IFP size
when redundancy mode is in use for error correction.
3) When an IFP larger than the calculated 'far max IFP' size is presented for
writing, truncate it rather than putting in the buffer and allowing the buffer
to overflow; this will cause the ends to retrain to a lower bit rate that
produces IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to the one endpoint
supplying a T38FaxMaxDatagram value that is improperly calculated and is
too low to be of use; we have configuration options available to override
this behavior.
4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer
needed.
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r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines
Merged revisions 222878 via svnmerge from
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r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines
Make filestream frame handling safer by isolating frames before returning them.
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/
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r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines
Recorded merge of revisions 222152 via svnmerge from
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
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/
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r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
Merged revisions 221086 via svnmerge from
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r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
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/
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r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines
Merged revisions 219136 via svnmerge from
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r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines
Prevent a potential race condition and crash when hanging up a channel by 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/
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r217074 | kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 lines
Ensure that the default autoconf CFLAGS are not used.
A recent change to the configure script that allows the user to specify
CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build
system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That
problem is now corrected.
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r214696 | kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9 lines
Ensure that CFLAGS and/or LDFLAGS provided to configure script are preserved.
Cross-compilation environments want to provide 'defaults' for compiler and
linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the
environment or as command-line arguments to the configure script. This patch
modifies the configure script and Makefile to preserve these settings and
ensure they are used in the build process.
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r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines
Resolve a T.38 negotiation issue left over from the udptl-updates merge.
The udptl-updates branch that was merged yesterday failed to properly send back
T.38 SDP responses with the correct error correction mode, if the incoming SDP
from the other end caused us to change error correction modes. This patch
corrects that situation.
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r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
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r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines
Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).
This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.
(closes issue #14849)
Reported by: afosorio
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r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines
Move OpenSSL initialization to a single place, make library usage thread-safe.
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
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This patch adds reference counting for sip dialogs into 1.6.0.
When proc_session_timer() is called from the scheduler thread
it has no guarantee the session timer's dialog won't be freed
from underneath it. Now the session timer holds a reference
to the dialog, preventing it from being destroyed during the
middle of proc_session_timer().
(closes issue #13623)
Reported by: Nik Soggia
Review: https://reviewboard.asterisk.org/r/302/
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r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines
Merged revisions 204681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds
a cli command, core show device2extenstate, to display all possible
state mappings.
(closes issue #15413)
Reported by: legart
Patches:
exten_helper.diff uploaded by dvossel (license 671)
Tested by: dvossel, legart, amilcar
Review: https://reviewboard.asterisk.org/r/301/
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r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines
Merged revisions 201261 via svnmerge from
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r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines
Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.
When the list to be appended is empty, and the list to be appended to is *not*,
AST_LIST_APPEND_LIST would actually cause the target list to become broken,
and no longer have a pointer to its last entry. This patch fixes the problem.
(reported by Stanislaw Pitucha on the asterisk-dev mailing list)
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r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines
Merged revisions 200991 via svnmerge from
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
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r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines
Another minor fix to compiler attribute checking.
Defaulting to 'static' for the function scope was bad... so remove it.
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r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines
Ensure that configure-script testing for compiler attributes actually works.
The configure script tests for compiler attributes didn't actually enable
enough warnings or provide a proper test harness to determine whether the
compiler supports the attribute in question or not; this caused gcc 4.1 to
report that it supports 'weakref', but it doesn't actually support it in the
way that is needed for our optional API mechanism. The new configure script
test will properly distinguish between full support and partial support
for this attribute, among others.
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r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines
Merged revisions 199626,199628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines
Increase the size of our thread stack on 64 bit processors.
We were setting the stack size for each thread to 240KB regardless of
architecture, which meant that in some scenarios we actually had less available
stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we
calculate the stack size we reserve based on the platform's __WORDSIZE, which
gives us:
32 bit -> 240KB
64 bit -> 496KB
128 bit -> 1008KB (that's right, we're ready for 128 bit processors)
Patch typed by me but written by several members of #asterisk-dev, including
Kevin, Tilghman, and Qwell.
(closes issue #14932)
Reported by: jpiszcz
Patches:
06052009_issue14932.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines
Fix a typo in the stack size calculation just introduced.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@199632 65c4cc65-6c06-0410-ace0-fbb531ad65f3
