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(closes issue #15757)
Reported by: Marquis
Patches:
distributed_devstate-XMPP.txt uploaded by lmadsen (license 10)
Tested by: Marquis, lmadsen, marcelloceschia
Review: https://reviewboard.asterisk.org/r/351/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270442 | lmadsen | 2010-06-15 07:47:03 -0500 (Tue, 15 Jun 2010) | 1 line
Move information about zonemessages into the [zonemessages] section.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270331 | pabelanger | 2010-06-14 17:31:59 -0400 (Mon, 14 Jun 2010) | 14 lines
Properly play first file in sort list.
When using sort=alpha we would always skip the first file
in the list first time through. We now check for that
properly.
(closes issue #17470)
Reported by: pabelanger
Patches:
sort.aplha.patch uploaded by pabelanger (license 224)
Tested by: lmadsen
Review: https://reviewboard.asterisk.org/r/703/
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Also found a place where sig_pri_init_pri() was inlined and called it
instead.
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The "channel" option would chop the channel name at the last '-', which made
it useless for something like a channel transfer from the dialplan. The
"fullchannel" option will return the channel name as-is.
ABE-2218
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Add the append_msn_to_cid_tag option to chan_dahdi like chan_misdn.
Review: https://reviewboard.asterisk.org/r/696/
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Theoretically the ./configure script is a pure bourne-shell script.
Practically it may be run by bash if /bin/sh is not good enough. But we should not count on it. See bug report for the gory details.
(closes issue #17485)
Patches:
0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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FreeBSD / OSX builds.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270078 | pabelanger | 2010-06-12 14:54:20 -0400 (Sat, 12 Jun 2010) | 2 lines
Fix typo in example
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This way the libraries can be found even if they are in
non-standard locations.
(closes issue #16155)
Reported by: jcollie
Patches:
0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch uploaded by jcollie (license 412)
Tested by: jsmith, tilghman, pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r269960 | tilghman | 2010-06-11 13:23:05 -0500 (Fri, 11 Jun 2010) | 8 lines
For SpeeX, 0 bits remaining is valid and does not need an emitted warning.
(closes issue #15762)
Reported by: nblasgen
Patches:
issue15672.patch uploaded by pabelanger (license 224)
Tested by: nblasgen
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(closes issue #16965)
Reported by: rrb3942
Patches:
DBGetComplete.patch uploaded by rrb3942 (license 1003)
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(closes issue #17031)
Reported by: pabelanger
Patches:
Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
Tested by: pabelanger, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r269821 | mmichelson | 2010-06-10 14:30:12 -0500 (Thu, 10 Jun 2010) | 19 lines
Fix potential crash when writing raw SLIN audio on a PLC-enabled channel.
The issue here was that the frame created when adjusting for PLC had no offset
to its audio data. If this frame were translated to another format prior to
being sent out an RTP socket, all went well because the translation code would
put an appropriate offset into the frame. However, if the SLIN audio were not
translated before being sent out the RTP socket, bad things would happen.
Specifically, the ast_rtp_raw_write makes the assumption that the frame has
at least enough of an offset that it can accommodate an RTP header. This was
not the case. As such, data was being written prior to the allocation, likely
corrupting the data the memory allocator had written. Thus when the time came
to free the data, all hell broke loose. ....Well, Asterisk crashed at least.
The fix was just what one would expect. Offset the data in the frame by a reasonable
amount. The method I used is a bit odd since the data in the frame is 16 bit integers
and not bytes. I left a big ol' comment about it. This can be improved on if someone
is interested. I was more interested in getting the crash resolved.
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Review: https://reviewboard.asterisk.org/r/688/
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logger.conf.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r269635 | tilghman | 2010-06-10 02:52:34 -0500 (Thu, 10 Jun 2010) | 9 lines
Ensure restartable system calls can restart (BSD signal semantics).
This eliminates the annoying <beep> on the console.
(closes issue #17477)
Reported by: jvandal
Patches:
20100610__issue17477.diff.txt uploaded by tilghman (license 14)
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http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r269495 | russell | 2010-06-09 17:18:37 -0500 (Wed, 09 Jun 2010) | 2 lines
Don't stop Asterisk if chan_oss fails to register 'Console' (due to another channel driver already claiming it).
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r269426 | qwell | 2010-06-09 16:19:17 -0500 (Wed, 09 Jun 2010) | 6 lines
Let systems without a working fork() use res_musiconhold.
Files mode doesn't require anything special, so that can still be used just fine.
AST-357
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Valgrind pointed out that attempting to get an IE value from an event that has
no IEs produces an invalid memory read past the end of the event. Thanks to
mmichelson for pointing the problem out to me and then testing the fix.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r269334 | pabelanger | 2010-06-09 13:24:53 -0400 (Wed, 09 Jun 2010) | 12 lines
Fix Debian init script to not use -c.
When using the init script as-is currently, it could cause issues on Debian
such as high CPU usage. This fix has worked for several people so I'm
implementing the change. We now handle color displays properly.
(closes issue #16784)
Reported by: pabelanger
Patches:
20100530__issue16784__2.diff.txt uploaded by tilghman (license 14)
Tested by: pabelanger, tilghman
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Calling dahdi_indicate() within dahdi_fixup() while the owner pointers are
in a potentially inconsistent state is a potentially bad thing in
principle.
However, calling dahdi_indicate() when the channel private lock is already
held can cause a deadlock if the PRI lock is needed because
dahdi_indicate() will also get the channel private lock. The pri_grab()
function assumes that the channel private lock is held once to avoid
deadlock.
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The result for moh_register was not verified to guarantee
the mohclass as added to the container.
(closes issue #16993)
Reported by: dmitri
Patches:
res_musiconhold_rtclass2.patch uploaded by dmitri (license 1001)
moh_crash2.diff uploaded by dvossel (license 671)
Tested by: dmitri
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* chan_dahdi supports dialing configuring and dialing by device file name.
DAHDI/span-name!local!1 will use /dev/dahdi/span-name/local/1 . Likewise
it may appear in chan_dahdi.conf as 'channel => span-name!local!1'.
* A new options for chan_dahdi.conf: 'ignore_failed_channels'. Boolean.
False by default. If set, chan_dahdi will ignore failed 'channel' entries.
Handy for the above name-based syntax as it does not depend on
initialization order.
* have my_pri_make_cc_dialstring() only manupulate dial-strings of group
(gGrR) dialing, which make it lsightly more complicated.
https://reviewboard.asterisk.org/r/535/
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unpackaged dependencies (such as NBS and libresample).
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People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.
Patch by snuffy.
(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy
Review: https://reviewboard.asterisk.org/r/461/
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(closes issue #17087)
Reported by: bklang
Patches:
app-fax-null-sprintf1.diff uploaded by mnicholson (license 96)
Tested by: bklang
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #16685)
Reported by: pprindeville
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r269006 | seanbright | 2010-06-08 11:28:49 -0400 (Tue, 08 Jun 2010) | 11 lines
Reduce startup time for cdr_tds with large CDR tables.
Since we are just checking for table existence, add a WHERE clause that will
return no rows but will raise an error if the table doesn't exist.
(closes issue #17380)
Reported by: kkwong
Patches:
issue17380-01.patch uploaded by seanbright (license 71)
Tested by: kkwong
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Update note in sip.conf.sample about externip and externhost with STUN.
(closes issue #16323)
Reported by: klaus3000
Patches:
sip.conf.sample-patch.txt uploaded by klaus3000 (license 65)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #17336)
Reported by: snuffy
Patches:
doxygen-fixes1.diff uploaded by snuffy (license 35)
Tested by: russell
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After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
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