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r229499 | dbrooks | 2009-11-11 13:48:18 -0600 (Wed, 11 Nov 2009) | 15 lines
Merged revisions 229498 via svnmerge from
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r229498 | dbrooks | 2009-11-11 13:46:19 -0600 (Wed, 11 Nov 2009) | 8 lines
Solaris doesn't like NULL going to ast_log
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
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r229361 | tilghman | 2009-11-10 16:14:22 -0600 (Tue, 10 Nov 2009) | 19 lines
Merged revisions 229360 via svnmerge from
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r229360 | tilghman | 2009-11-10 16:09:16 -0600 (Tue, 10 Nov 2009) | 12 lines
If two pattern classes start with the same digit and have the same number of 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
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r228897 | lmadsen | 2009-11-09 09:38:38 -0600 (Mon, 09 Nov 2009) | 14 lines
Merged revisions 228896 via svnmerge from
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r228896 | lmadsen | 2009-11-09 09:37:43 -0600 (Mon, 09 Nov 2009) | 6 lines
Update WARNING message.
Update a WARNING message to give a suggested fix when encountered.
(closes issue #16198)
Reported by: atis
Tested by: atis
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r228693 | dvossel | 2009-11-06 16:35:44 -0600 (Fri, 06 Nov 2009) | 16 lines
Merged revisions 228692 via svnmerge from
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r228692 | dvossel | 2009-11-06 16:33:27 -0600 (Fri, 06 Nov 2009) | 9 lines
fixes audiohook write crash occuring in chan_spy whisper mode.
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
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r228620 | mnicholson | 2009-11-06 13:47:11 -0600 (Fri, 06 Nov 2009) | 15 lines
Merged revisions 228378 via svnmerge from
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r228378 | mnicholson | 2009-11-06 10:26:59 -0600 (Fri, 06 Nov 2009) | 8 lines
Properly handle '=' while decoding base64 messages and null terminate strings returned from BASE64_DECODE.
(closes issue #15271)
Reported by: chappell
Patches:
base64_fix.patch uploaded by chappell (license 8)
Tested by: kobaz
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r228410 | file | 2009-11-06 12:42:23 -0400 (Fri, 06 Nov 2009) | 14 lines
Merged revisions 228409 via svnmerge from
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r228409 | file | 2009-11-06 12:41:20 -0400 (Fri, 06 Nov 2009) | 7 lines
Fix a bug caused by a partially invalid frame (from the jitterbuffer) passing through the Asterisk core.
(closes issue #15560)
Reported by: jvandal
(closes issue #15709)
Reported by: covici
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r228339 | dvossel | 2009-11-06 09:42:46 -0600 (Fri, 06 Nov 2009) | 12 lines
Merged revisions 228338 via svnmerge from
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r228338 | dvossel | 2009-11-06 09:41:41 -0600 (Fri, 06 Nov 2009) | 5 lines
fixes crash in astfd.c
(closes issue #15981)
Reported by: slavon
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r227372 | qwell | 2009-11-03 13:59:46 -0600 (Tue, 03 Nov 2009) | 9 lines
Fix some build issues on Solaris.
(closes issue #14517)
(SWP-109)
Reported by: asgaroth
Patches:
bug_14517.diff uploaded by snuffy (license 35)
Tested by: asgaroth, snuffy, dougm, qwell
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r226159 | tilghman | 2009-10-27 15:22:07 -0500 (Tue, 27 Oct 2009) | 14 lines
Merged revisions 226138 via svnmerge from
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r226138 | tilghman | 2009-10-27 15:16:49 -0500 (Tue, 27 Oct 2009) | 7 lines
Manager output is not always NULL-terminated, so force a NULL at the end of the filestream.
(closes issue #15495)
Reported by: pdf
Patches:
20090916__issue15495.diff.txt uploaded by tilghman (license 14)
Tested by: pdf
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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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r224856 | tilghman | 2009-10-20 17:09:07 -0500 (Tue, 20 Oct 2009) | 12 lines
Merged revisions 224855 via svnmerge from
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r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines
Pay attention to the return value of the manipulate function.
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).
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r224774 | file | 2009-10-20 14:47:34 -0300 (Tue, 20 Oct 2009) | 12 lines
Merged revisions 224773 via svnmerge from
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r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines
Add support for relaying early media in the features attended transfer option.
(closes issue #14828)
Reported by: licedey
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r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines
Merged revisions 224670 via svnmerge from
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r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines
Correct timestamp calculations when RTP sample rates over 8kHz are used.
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.
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r223487 | russell | 2009-10-11 12:25:42 -0500 (Sun, 11 Oct 2009) | 17 lines
Merged revisions 223485-223486 via svnmerge from
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r223485 | russell | 2009-10-11 12:22:52 -0500 (Sun, 11 Oct 2009) | 6 lines
Don't use data outside of its scope.
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.
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r223486 | russell | 2009-10-11 12:25:06 -0500 (Sun, 11 Oct 2009) | 2 lines
Remove some unnecessary code.
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r223273 | mnicholson | 2009-10-09 13:34:08 -0500 (Fri, 09 Oct 2009) | 14 lines
Merged revisions 223225 via svnmerge from
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r223225 | mnicholson | 2009-10-09 13:20:11 -0500 (Fri, 09 Oct 2009) | 8 lines
Signal timeouts by returning AST_CONTROL_RINGING when originating calls.
(closes issue #15104)
Reported by: nblasgen
Patches:
manager-timeout1.diff uploaded by mnicholson (license 96)
Tested by: nblasgen, mnicholson
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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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This looks like it was just missed during a merge.
(closes issue #15841)
Reported by: amorsen
Patches:
ast_devstate_aggregate_init-in-ast_extension_state2.patch uploaded by amorsen (license 676)
Tested by: amorsen
(closes issue #15852)
Reported by: amorsen
Tested by: amorsen, farisraouf
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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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r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines
Allow non-compliant T.38 endpoints to be supportable via configuration option.
Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.
This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.
In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).
(issue #15586)
Reported by: globalnetinc
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r221971 | tilghman | 2009-10-02 11:59:57 -0500 (Fri, 02 Oct 2009) | 9 lines
Merged revisions 221970 via svnmerge from
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r221970 | tilghman | 2009-10-02 11:58:03 -0500 (Fri, 02 Oct 2009) | 2 lines
Ensure the result of the hash function is positive. Negative array offsets suck.
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r221920 | tilghman | 2009-10-01 22:04:34 -0500 (Thu, 01 Oct 2009) | 4 lines
Initialize a variable that we check immediately upon startup.
(closes issue #15973)
Reported by: atis
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r221777 | tilghman | 2009-10-01 18:59:15 -0500 (Thu, 01 Oct 2009) | 9 lines
Merged revisions 221776 via svnmerge from
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r221776 | tilghman | 2009-10-01 18:53:12 -0500 (Thu, 01 Oct 2009) | 2 lines
Fix a bunch of off-by-one errors
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r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines
Remove ability to control T.38 FAX error correction from udptl.conf.
chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer
(or global) basis for a couple of releases now, which is where it should have been
all along. This patch removes the ability to configure it in udptl.conf, but issues
a warning if the user tries to do, telling them to look at sip.conf.sample for how
to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is
already a default for FEC error correction even if the user does not specify any mode,
so this change will not turn off error correction by default, it will have the same
default value that has been in the udptl.conf sample file.
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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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r221201 | tilghman | 2009-09-30 11:56:42 -0500 (Wed, 30 Sep 2009) | 14 lines
Merged revisions 221200 via svnmerge from
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r221200 | tilghman | 2009-09-30 11:55:21 -0500 (Wed, 30 Sep 2009) | 7 lines
Avoid a potential NULL dereference.
(closes issue #15865)
Reported by: kobaz
Patches:
20090915__issue15865.diff.txt uploaded by tilghman (license 14)
Tested by: kobaz
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r220365 | dvossel | 2009-09-24 15:37:20 -0500 (Thu, 24 Sep 2009) | 8 lines
fixes tcptls_session memory leak caused by ref count error
(closes issue #15939)
Reported by: dvossel
Review: https://reviewboard.asterisk.org/r/375/
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r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines
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r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines
Implicitly sending a progress signal breaks some applications.
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
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r219654 | tilghman | 2009-09-20 12:55:49 -0500 (Sun, 20 Sep 2009) | 15 lines
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r219653 | tilghman | 2009-09-20 12:52:05 -0500 (Sun, 20 Sep 2009) | 8 lines
Really stop the stream, when ast_closestream() is called.
(closes issue #15129)
Reported by: bmh
Patches:
20090918__issue15129.diff.txt uploaded by tilghman (license 14)
Review:
https://reviewboard.asterisk.org/r/372/
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r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines
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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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r219061 | tilghman | 2009-09-16 18:42:12 -0500 (Wed, 16 Sep 2009) | 15 lines
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r219023 | tilghman | 2009-09-16 18:21:53 -0500 (Wed, 16 Sep 2009) | 8 lines
Properly deal with quotes in the arguments of '#exec' includes.
(closes issue #15583)
Reported by: pkempgen
Patches:
20090726__issue15583.diff.txt uploaded by tilghman (license 14)
20090726__issue15583-1.4-4.diff.txt uploaded by pkempgen (license 169)
Tested by: pkempgen
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r218868 | dbrooks | 2009-09-16 13:06:42 -0500 (Wed, 16 Sep 2009) | 20 lines
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r218867 | dbrooks | 2009-09-16 13:00:45 -0500 (Wed, 16 Sep 2009) | 13 lines
Fixes CID pattern matching behavior to mirror that of extension pattern matching.
Pattern matching for extensions uses a type of scoring system, giving values for
specificity to each character in the pattern. Unfortunately, this is done character
by character, in order. This does lead to some less specific patterns being first
in line for matching, but it will usually get the job done.
This patch merely brings CID matching to the same level as extension matching.
This patch does not attempt to tackle the problem shared by extension matching.
(closes issue #14708)
Reported by: klaus3000
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Since 1.6.X still has the deprecated 'rtp debug ip <foo>'
this patch is different from the fix that went into trunk
(closes issue #15711)
Reported by: davidw
Patches:
2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
Tested by: davidw
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r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines
Merged revisions 216430 via svnmerge from
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r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
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r216547 | tilghman | 2009-09-04 12:31:44 -0500 (Fri, 04 Sep 2009) | 3 lines
Enable turning off the application delimiter warning with the 'dontwarn' option.
Suggested on the -dev list, and implemented in an alternate way by me.
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r216506 | mvanbaak | 2009-09-04 17:05:05 +0200 (Fri, 04 Sep 2009) | 9 lines
Merged revisions 216435 via svnmerge from
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r216435 | mvanbaak | 2009-09-04 15:56:10 +0200 (Fri, 04 Sep 2009) | 2 lines
make asterisk compile under devmode with DEBUG_THREADS enabled on OpenBSD
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r216222 | mvanbaak | 2009-09-04 08:08:33 +0200 (Fri, 04 Sep 2009) | 3 lines
make sure 'start' is always initialized.
Makes asterisk compile with --enable-dev-mode
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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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r215161 | kpfleming | 2009-09-01 14:50:48 -0500 (Tue, 01 Sep 2009) | 3 lines
Ensure that frame dumps of AST_CONTROL_T38_PARAMETERS frames are properly
decoded.
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r214702 | tilghman | 2009-08-28 15:14:39 -0500 (Fri, 28 Aug 2009) | 15 lines
Merged revisions 214701 via svnmerge from
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r214701 | tilghman | 2009-08-28 15:13:32 -0500 (Fri, 28 Aug 2009) | 8 lines
Modify comment to be a bit more accurate.
We have kept this comment around long enough, that it's pretty clear that we're
keeping the code, because changing the code would require a pretty fundamental
architectural shift. We've also taken criticism in some quarters, because it
was believed that it was referring to the code being nasty. No, the code isn't
nasty, just the operation itself is rather odd. Fixed for eternity (probably
not).
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r214195 | dvossel | 2009-08-26 11:38:53 -0500 (Wed, 26 Aug 2009) | 25 lines
Merged revisions 214194 via svnmerge from
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r214194 | dvossel | 2009-08-26 11:36:42 -0500 (Wed, 26 Aug 2009) | 19 lines
ast_write() ignores ast_audiohook_write() results
In ast_write(), if a channel has a list of audiohooks, those
lists are written to and the resulting frame is what ast_write()
should continue with. The problem was the returned audiohook frame
was not being handled at all, and the original frame passed
into it did not contain the mixed audio, so essentially audio
was being lost. One result of this was chan_spy's whisper
mode no longer worked. To complicate the issue, frames
passed into ast_write may either be a single frame, or a list
of frames. So, as the list of frames is processed in the
audiohook_write, the returned frames had to be added to a new
list.
(closes issue #15660)
Reported by: corruptor
Tested by: dvossel
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r214071 | tilghman | 2009-08-25 14:32:48 -0500 (Tue, 25 Aug 2009) | 17 lines
Merged revisions 214068-214069 via svnmerge from
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r214068 | tilghman | 2009-08-25 14:26:50 -0500 (Tue, 25 Aug 2009) | 6 lines
Fix pronunciation of German dates.
(closes issue #15273)
Reported by: Benjamin Kluck
Patches:
say_c.patch uploaded by Benjamin Kluck (license 803)
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r214069 | tilghman | 2009-08-25 14:28:42 -0500 (Tue, 25 Aug 2009) | 2 lines
I should always compile before committing...
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r213971 | tilghman | 2009-08-25 01:35:37 -0500 (Tue, 25 Aug 2009) | 14 lines
Merged revisions 213970 via svnmerge from
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r213970 | tilghman | 2009-08-25 01:34:44 -0500 (Tue, 25 Aug 2009) | 7 lines
Improve error message by informing user exactly which function is missing a parethesis.
(closes issue #15242)
Reported by: Nick_Lewis
Patches:
pbx.c-funcparenthesis.patch2 uploaded by dbrooks (license 790)
pbx.c-funcparenthesis-1.4.diff uploaded by loloski (license 68)
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r213327 | mnicholson | 2009-08-20 15:29:32 -0500 (Thu, 20 Aug 2009) | 7 lines
Fix a crash by checking the proper pointer for validity before deferencing it.
(closes issue #15751)
Reported by: atis
Patches:
ast_bridge_call_peer_cdr.patch uploaded by atis (license 242)
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r213179 | qwell | 2009-08-19 17:38:46 -0500 (Wed, 19 Aug 2009) | 5 lines
Fix compile when certain G711 menuselect options are enabled.
(closes issue #15697)
Reported by: slavon
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r212574 | seanbright | 2009-08-17 14:18:16 -0400 (Mon, 17 Aug 2009) | 8 lines
Correct the return value check for ast_safe_system.
The logic here was reversed as ast_safe_system returns -1 on error and not on
success. Fix suggested by reporter.
(closes issue #15667)
Reported by: loic
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r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009) | 9 lines
Merged revisions 211274 via svnmerge from
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r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009) | 2 lines
Small oops. Clear the flags which have been checked.
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r210992 | kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13 lines
Workaround broken T.38 endpoints that offer tiny MaxDatagram sizes.
Some T.38 endpoints treat T38FaxMaxDatagram as the maximum IFP size that should
be sent to them, rather than the maximum packet payload size. If such an
endpoint also requests UDPRedundancy as the error correction mode, we'll end
up calculating a tiny maximum IFP size, so small as to be unusable. This patch
sets a lower bound on what we'll consider the remote's maximum IFP size to be,
assuming that endpoints that do this really can accept larger packets than
they've offered to accept.
(closes issue #15649)
Reported by: dazza76
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