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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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r235573 | tilghman | 2009-12-18 15:19:43 -0600 (Fri, 18 Dec 2009) | 9 lines
Merged revisions 235572 via svnmerge from
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r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) | 2 lines
Point to the typical missing package, not the cryptic "termcap support".
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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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r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines
Merged revisions 231614 via svnmerge from
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r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines
Remove duplicate entries from voicemail format lists. This prevents 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/
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r230583 | dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines
audiohook signal trigger on every status change
(issue #14618)
Review: https://reviewboard.asterisk.org/r/434/
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r228858 | mnicholson | 2009-11-09 08:37:07 -0600 (Mon, 09 Nov 2009) | 15 lines
Merged revisions 228827 via svnmerge from
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r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov 2009) | 8 lines
Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices.
(closes issue #15588)
Reported by: zerohalo
Patches:
20090820__issue15588.diff.txt uploaded by tilghman (license 14)
Tested by: zerohalo
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r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines
Merged revisions 226304 via svnmerge from
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r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines
Fix documentation (pointed out by TheDavidFactor on #-dev)
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r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines
Merged revisions 225105 via svnmerge from
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r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines
Fix documentation for ast_softhangup() and correct the misuse thereof.
(closes issue #16103)
Reported by: majorbloodnok
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r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009) | 12 lines
Merged revisions 224931 via svnmerge from
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r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines
Isolate frames returned from a DSP instance or codec translator.
The reasoning for these changes are the same as what I wrote in the commit
message for rev 222878.
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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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r222873 | dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines
fixes an ast_netsock_list memory leak.
ABE-1998
Review: https://reviewboard.asterisk.org/r/395/
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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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r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009) | 2 lines
Allow AES to compile, when OpenSSL is not present.
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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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r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines
Merge code associated with AST-2009-006
(closes issue #12912)
Reported by: rathaus
Tested by: tilghman, russell, dvossel, dbrooks
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r214819 | tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines
If lua is detected with the lua5.1 prefix (or not), adjust the include path accordingly.
Based upon feedback to a release announcement on the -users list. See
http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html
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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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r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009) | 14 lines
Merged revisions 214517 via svnmerge from
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r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) | 7 lines
Use autoconf to detect libcurl, as this enables cross-compilation checks, something we didn't allow before.
(closes issue #15714)
Reported by: pprindeville
Patches:
20090813__issue15714.diff.txt uploaded by tilghman (license 14)
Tested by: pprindeville
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r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009) | 9 lines
Merged revisions 214436 via svnmerge from
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r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) | 2 lines
One more build system change, to make the descriptions look better, if we have better information.
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r214360 | tilghman | 2009-08-27 11:12:03 -0500 (Thu, 27 Aug 2009) | 10 lines
Merged revisions 214357 via svnmerge from
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r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) | 3 lines
Make autoheader descriptions render correctly in our autoconfig.h file.
(Figured out while working with issue #14906)
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r214152 | tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines
Not all versions of gnu-linux use glibc, which contains iconv. Some (especially embedded systems) don't have iconv at all.
(closes issue #15169)
Reported by: pprindeville
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r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009) | 14 lines
Merged revisions 213559 via svnmerge from
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r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) | 7 lines
Permit DEBUG_FD_LEAKS to be used with C++ source files.
(closes issue #15698)
Reported by: slavon
Patches:
20090817__issue15698.diff.txt uploaded by tilghman (license 14)
Tested by: slavon, tilghman
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r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
Allow Gosub to recognize quote delimiters without consuming them.
(closes issue #15557)
Reported by: rain
Patches:
20090723__issue15557.diff.txt uploaded by tilghman (license 14)
Tested by: rain
Review: https://reviewboard.asterisk.org/r/316/
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r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
Fixes numerous spelling errors. Patch submitted by alecdavis.
(closes issue #15595)
Reported by: alecdavis
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r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize"
(closes issue #15571)
Reported by: alecdavis
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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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r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines
Merged revisions 205599 via svnmerge from
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r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines
Changing ast_samp2tv to not use floating point.
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r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines
Merged revisions 205471 via svnmerge from
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r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines
Fixes 8khz assumptions
Many calculations assume 8khz is the codec rate. This
is not always the case. This patch only addresses chan_iax.c
and res_rtp_asterisk.c, but I am sure there are other areas
that make this assumption as well.
Review: https://reviewboard.asterisk.org/r/306/
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r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines
Merged revisions 205409 via svnmerge from
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r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines
moving ast_devstate_to_extenstate to pbx.c from devicestate.c
ast_devstate_to_extenstate belongs in pbx.c. This change
fixes a compile time error with chan_vpb as well.
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r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines
Merged revisions 205215 via svnmerge from
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r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines
ast_samp2tv needs floating point for 16khz audio
In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000.
The .5 is currently stripped off because we don't calculate
using floating points. This causes madness with 16khz audio.
(issue ABE-1899)
Review: https://reviewboard.asterisk.org/r/305/
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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
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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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r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines
Improve T.38 negotiation by exchanging session parameters between application and channel.
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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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r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines
Merged revisions 199856 via svnmerge from
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r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines
__WORDSIZE is not available on all platforms, so use sizeof(void *) instead.
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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
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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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r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines
Merged revisions 199022 via svnmerge from
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r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
Safely handle AMI connections/reload requests that occur during startup.
During asterisk startup, a lock on the list of modules is obtained by the
primary thread while each module is initialized. Issue 13778 pointed out a
problem with this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a connected client
(via Action: Command), causing a deadlock.
The resolution for 13778 was to move initialization of the manager to happen
after the other modules had already been lodaded. While this fixed this
particular issue, it caused a problem for users (like FreePBX) who call AMI
scripts via an #exec in a configuration file (See issue 15189).
The solution I have come up with is to defer any reload requests that come in
until after the server is fully booted. When a call comes in to
ast_module_reload (from wherever) before we are fully booted, the request is
added to a queue of pending requests. Once we are done booting up, we then
execute these deferred requests in turn.
Note that I have tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded, and if a
general reload request comes in ('module reload') the queue is flushed and we
only issue a single deferred reload for the entire system.
As for how this will impact existing installations - Before 13778, a reload
issued before module initialization was completed would result in a deadlock.
After 13778, you simply couldn't connect to the manager during startup (which
causes problems with #exec-that-calls-AMI configuration files). I believe this
is a good general purpose solution that won't negatively impact existing
installations.
(closes issue #15189)
(closes issue #13778)
Reported by: p_lindheimer
Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer, seanbright
Review: https://reviewboard.asterisk.org/r/272/
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r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
Generic call forward api, ast_call_forward()
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
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r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines
Merged revisions 198068 via svnmerge from
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r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
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r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines
Recorded merge of revisions 197588 via svnmerge from
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r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
Allow for media to arrive from an alternate source when responding to a reinvite with 491.
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
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r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines
Merged revisions 197537 via svnmerge from
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r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
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