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r228441 | dvossel | 2009-11-06 11:22:31 -0600 (Fri, 06 Nov 2009) | 3 lines
Fixes merging issue from 1.4, frame data is held in data.ptr in trunk
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r228420 | dvossel | 2009-11-06 11:09:01 -0600 (Fri, 06 Nov 2009) | 19 lines
Merged revisions 228418 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r228418 | dvossel | 2009-11-06 11:07:13 -0600 (Fri, 06 Nov 2009) | 13 lines
fixes segfault in iLBC
For reasons not yet known, it appears possible for an ast_frame
to have a datalen greater than zero while the actual data is NULL
during Packet Loss Concealment. Most codecs don't support PLC so
this doesn't affect them. This patch catches the malformed frame
and prevents the crash from occuring. Additional efforts to determine
why it is possible for a frame to look like this are still being
investigated.
(issue #16979)
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r228189 | jpeeler | 2009-11-05 15:23:06 -0600 (Thu, 05 Nov 2009) | 11 lines
Fix the fix for chanspy option o
In 224178, I assumed the uploaded patch was correct as it had received positive
feedback. The flags were being checked in the incorrect location. Upon testing
the fix this time it was also found that the flags from the dialplan weren't
being copied to the chanspy_translation_helper.
(closes issue #16167)
Reported by: marhbere
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r228145 | dbrooks | 2009-11-05 13:34:50 -0600 (Thu, 05 Nov 2009) | 16 lines
Merged revisions 228078 via svnmerge from
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r228078 | dbrooks | 2009-11-05 12:59:41 -0600 (Thu, 05 Nov 2009) | 9 lines
chan_misdn Asterisk 1.4.27-rc2 crash
Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested
by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached
a full bt." This patch zeros out an ast_frame.
(closes issue #16041)
Reported by: francesco_r
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r227945 | jpeeler | 2009-11-04 17:50:59 -0600 (Wed, 04 Nov 2009) | 21 lines
Merged revisions 227944 via svnmerge from
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r227944 | jpeeler | 2009-11-04 17:47:08 -0600 (Wed, 04 Nov 2009) | 14 lines
Fix incorrect filename comparsion after monitor file change
The logic to detect if a requested file is indeed a different file from the
current file was incorrect. The main issue being confusion of the use of
filename_base which was previously set without pathing information and then
compared to another full path. Robust file comparison logic has been added
to properly check if two files are the same even if symlinks are used.
(closes issue #15313)
Reported by: caspy
Patches:
20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325)
but mostly tilghman's work
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With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future.
(closes issue #14994)
Reported by: frawd
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r227361 | lmadsen | 2009-11-03 13:25:18 -0600 (Tue, 03 Nov 2009) | 11 lines
Additional fixes to the extensions.conf.sample file.
Update the extensions.conf.sample [stdexten] context so that we use the
variable instead of requiring it to be passed explicitly. Also updated uses of
the [stdexten] context throughout.
(closes issue #15858)
Reported by: pprindeville
Patches:
stdexten-context-update.txt uploaded by lmadsen (license 10)
Tested by: pprindeville
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SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.
(closes issue #15152)
Reported by: palbrecht
Review: https://reviewboard.asterisk.org/r/420/
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SIP channel names were supposed to be unique by way of a name suffix derived from the
pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but
not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with
a simple incremented unsigned int.
(closes issue #15152)
Reported by: palbrecht
Review: https://reviewboard.asterisk.org/r/420/
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r226890 | file | 2009-11-02 14:08:54 -0400 (Mon, 02 Nov 2009) | 18 lines
Merged revisions 226889 via svnmerge from
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r226889 | file | 2009-11-02 14:08:11 -0400 (Mon, 02 Nov 2009) | 11 lines
Fix a bug where the recorded privacy introduction file would not get removed if the caller hung up
while the called party had not yet answered.
This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction
file under all scenarios. This was done to preserve the behavior that has existed for quite some time.
(closes issue #14674)
Reported by: ulogic
Patches:
bug14674.patch uploaded by jpeeler (license 325)
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r226384 | lmadsen | 2009-10-28 15:11:07 -0500 (Wed, 28 Oct 2009) | 17 lines
Merged revisions 226382 via svnmerge from
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r226382 | lmadsen | 2009-10-28 15:06:13 -0500 (Wed, 28 Oct 2009) | 9 lines
Update documentation in sip.conf.sample.
Update the documentation in sip.conf.sample in order to make it more clear
that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It
is only used to stop Asterisk from generating a reINVITE, but does not stop
it from accepting them if necessary.
(closes issue #15644)
Reported by: lmadsen
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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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* Set OSARCH to linux-gnu even if host_os is linux-gnueabi
* When checking if we are Linux, check OSARCH rather than host_os
The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than
'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case.
OSARCH is tested for the value of 'linux-gnu' in one or two places in the
tree. This patch also fixes the check libcap to check for $OSARCH rather
than $host_os .
See also: http://wiki.debian.org/ArmEabiPort
Merged revisions 225957 via svnmerge from
http://svn.digium.com/svn/asterisk/branches/1.4
Merged revisions 226018 via svnmerge from
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r225912 | jpeeler | 2009-10-26 14:40:26 -0500 (Mon, 26 Oct 2009) | 12 lines
ACL check not present for verifying SIP INVITEs
The ACL check in check_peer_ok was missing and has now been restored. The
missing check allowed for calls to be made on prohibited networks where an ACL
was defined in sip.conf and the allowguest option was set to off. See the AST
security advisory below for more information.
Merge code associated with AST-2009-007.
(closes issue #16091)
Reported by: thom4fun
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This backport resolves some issues handling audio frames during FAX processing,
and ensures that the FAX application doesn't accidentally get notified of a T.38
switchover at the end of a successful FAX.
(issue #16127)
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r225582 | kpfleming | 2009-10-23 09:02:42 -0500 (Fri, 23 Oct 2009) | 17 lines
Merged revisions 225581 via svnmerge from
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r225581 | kpfleming | 2009-10-23 09:00:01 -0500 (Fri, 23 Oct 2009) | 10 lines
Don't force menuselect.makeopts to be rebuilt on every build.
For some reason the menuselect.makeopts file was listed as PHONY in the Makefile,
resulting in 'make' needing to rebuild it for every build. This then resulted in
the embedded module rules being rebuilt on every build, which can be slow and is
unnecessary.
This patch fixes the problem by properly allowing 'make' to know when the
menuselect.makeopts file needs to be rebuilt (defining the proper dependencies).
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r225307 | dvossel | 2009-10-21 16:58:46 -0500 (Wed, 21 Oct 2009) | 20 lines
Merged revisions 225243 via svnmerge from
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r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines
IAX2: VNAK loop caused by signaling frames with no destination call number
It is possible for the PBX thread to queue up signaling frames before
a destination call number is received. This can result in signaling
frames being sent out with no destination call number. Since recent
versions of Asterisk require accurate destination callnumbers for all
Full Frames, this can cause a VNAK loop to occur. To resolve this
no signaling frames are sent until a destination callnumber is received,
and destination call numbers are now only required for iax_pvt matching
when the frame is an ACK.
Review: https://reviewboard.asterisk.org/r/413/
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r225033 | dvossel | 2009-10-21 09:39:10 -0500 (Wed, 21 Oct 2009) | 27 lines
Merged revisions 225032 via svnmerge from
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r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines
IAX/SIP shrinkcallerid option
The shrinking of caller id removes '(', ' ', ')', non-trailing '.',
and '-' from the string. This means values such as 555.5555 and
test-test result in 555555 and testtest. There are instances,
such as Skype integration, where a specific value is passed via
caller id that must be preserved unmodified. This patch makes
the shrinking of caller id optional in chan_sip and chan_iax in
order to support such cases. By default this option is on to
preserve previous expected behavior.
(closes issue #15940)
Reported by: dimas
Patches:
v2-15940.patch uploaded by dimas (license 88)
15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671)
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/408/
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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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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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r224331 | jpeeler | 2009-10-16 20:36:08 -0500 (Fri, 16 Oct 2009) | 20 lines
Merged revisions 224330 via svnmerge from
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r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines
Fix stale caller id data from being reported in AMI NewChannel event
The problem here is that chan_dahdi is designed in such a way to set
certain values in the dahdi_pvt only once. One of those such values
is the configured caller id data in chan_dahdi.conf. For PRI, the
configured caller id data could be overwritten during a call. Instead
of saving the data and restoring, it was decided that for all non-analog
channels it was simply best to not set the configured caller id in the
first place and also clear it at the end of the call.
(closes issue #15883)
Reported by: jsmith
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