[asterisk-commits] phsultan: branch phsultan/jingle-support r281182 - in /team/phsultan/jingle-s...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Aug 6 15:33:03 CDT 2010
Author: phsultan
Date: Fri Aug 6 15:31:52 2010
New Revision: 281182
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=281182
Log:
Merged revisions 268894,268896,268933,268969,268988,269007-269008,269083,269119,269153,269187,269196-269205,269238,269271,269307-269308,269346,269417,269486,269497,269569,269602,269636,269707,269711,269749,269822,269889,269936,269938,269976,270042,270079,270151,270184,270219,270260,270298,270332,270443,270519,270552,270584,270658,270660,270692,270726,270801,270836,270867,270936,270940,270974,270981,270983,270987,271056,271089,271124,271192,271231,271261-271262,271300,271336,271341,271483,271520,271551,271553-271554,271625,271657,271690,271762,271764,271831,271833,271867-271868,271903,271905,271977,272014,272052,272090,272109,272145-272146,272148,272150,272218,272243,272252,272254,272256-272257,272259-272260,272332,272368,272370,272447,272533,272557-272558,272568,272652,272684,272805,272880,272923,272926,272981,273054-273055,273058,273078,273142,273144,273197-273198,273233,273270,273312,273350,273352,273355,273427,273464,273522,273566,273641,273714-273715,273718,273830,273886,273982,274053,274094,274164,274243,274281,274284,274316,274418,274491-274492,274539-274540,274595,274639,274686,274727,274773,274782-274783,274785-274786,274827-274828,274866,274907,274909,274947,274984,275022,275028,275104-275105,275144,275147,275172,275186,275227,275249,275294,275307-275310,275312,275385,275424,275466-275467,275509,275551,275587,275626,275682,275725,275816,275863,275910,275995,275998,276074,276114,276118,276120,276122,276124,276127,276206,276219,276268,276347,276349,276389,276391-276393,276439,276441,276490,276493,276531,276570-276571,276616,276653,276731,276769,276788,276830,276869-276871,276908-276911,276950-276952,276989,277027-277028,277065,277102-277103,277143,277175,277183,277250,277262-277263,277331,277366,277409,277452,277467,277484,277488,277530,277657,277667,277703,277773,277775,277814,277837,277872-277873,277945,278024,278051,278096,278132,278168,278234,278272,278274-278275,278307,278361,278393,278425-278426,278461-278463,278465,278501,278536,278538-278539,278579,278619-278620,278708,278777,278809,278841,278873,278875,278908,278942-278945,278947,278957,278980,278982,278985,279063,279084,279115-279116,279118,279156,279245,279274,279285,279315,279391,279413,279443,279473,279503,279533,279564,279567,279569,279600,279602,279624,279659,279692,279725,279727,279756,279786,279816,279818,279851,279888,279917,279951,279954,280024,280059,280091,280093,280162,280196,280226,280234,280247,280270,280308,280340,280342,280344,280395,280415-280416,280447,280459,280518,280520,280550,280553,280555,280559,280589,280626-280627,280629,280673,280706-280707,280741,280743,280745,280779-280780,280810,280880,280910,280985,281054 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r268894 | twilson | 2010-06-08 07:29:08 +0200 (Tue, 08 Jun 2010) | 17 lines
Add SRTP support for Asterisk
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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r268896 | tilghman | 2010-06-08 08:16:43 +0200 (Tue, 08 Jun 2010) | 2 lines
Fix trunk build on Mac OS X.
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r268933 | tilghman | 2010-06-08 08:57:24 +0200 (Tue, 08 Jun 2010) | 2 lines
Release list lock before returning on error.
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r268969 | lmadsen | 2010-06-08 16:38:18 +0200 (Tue, 08 Jun 2010) | 7 lines
Fix some doxygen warnings.
(closes issue #17336)
Reported by: snuffy
Patches:
doxygen-fixes1.diff uploaded by snuffy (license 35)
Tested by: russell
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r268988 | lmadsen | 2010-06-08 17:23:20 +0200 (Tue, 08 Jun 2010) | 8 lines
Update note in sip.conf.sample.
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)
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r269007 | seanbright | 2010-06-08 17:39:52 +0200 (Tue, 08 Jun 2010) | 18 lines
Merged revisions 269006 via svnmerge from
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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r269008 | russell | 2010-06-08 17:41:23 +0200 (Tue, 08 Jun 2010) | 5 lines
Ensure CONFIG_FLAGS makes it into the build rules when doing out of tree builds.
(closes issue #16685)
Reported by: pprindeville
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r269083 | mnicholson | 2010-06-08 20:50:45 +0200 (Tue, 08 Jun 2010) | 9 lines
Don't pass null to manager_event()
(closes issue #17087)
Reported by: bklang
Patches:
app-fax-null-sprintf1.diff uploaded by mnicholson (license 96)
Tested by: bklang
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r269119 | tilghman | 2010-06-09 00:45:16 +0200 (Wed, 09 Jun 2010) | 2 lines
Fix build on Mac OS X (and maybe FreeBSD, too)
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r269153 | snuffy | 2010-06-09 01:48:17 +0200 (Wed, 09 Jun 2010) | 11 lines
Add High Resolution Times to CDRs for Asterisk
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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r269187 | russell | 2010-06-09 12:18:24 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libgtk2.0-dev to the packages list for install_prereq.
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r269196 | russell | 2010-06-09 12:21:23 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libmysqlclient-dev to install_prereq.
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r269197 | russell | 2010-06-09 12:23:05 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libbluetooth-dev to install_prereq.
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r269198 | russell | 2010-06-09 12:28:27 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libradiusclient-ng-dev to install_prereq.
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r269199 | russell | 2010-06-09 12:30:32 +0200 (Wed, 09 Jun 2010) | 2 lines
Add freetds-dev to install_prereq.
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r269200 | russell | 2010-06-09 12:33:32 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libcurl to install_prereq.
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r269201 | russell | 2010-06-09 12:45:10 +0200 (Wed, 09 Jun 2010) | 2 lines
Add an "install-unpackaged" command to install_prereq for installing unpackaged dependencies (such as NBS and libresample).
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r269202 | russell | 2010-06-09 12:47:19 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libopenais-dev to install_prereq.
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r269203 | russell | 2010-06-09 12:48:29 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libnewt-dev to install-prereq.
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r269204 | russell | 2010-06-09 12:53:26 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libpopt-dev, libical-dev, and libspandsp-dev to install_prereq.
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r269205 | russell | 2010-06-09 12:55:07 +0200 (Wed, 09 Jun 2010) | 2 lines
Add libjack-dev to install_prereq.
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r269238 | tzafrir | 2010-06-09 15:17:43 +0200 (Wed, 09 Jun 2010) | 14 lines
dial by name in chan_dahdi
* 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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r269271 | dvossel | 2010-06-09 17:09:25 +0200 (Wed, 09 Jun 2010) | 15 lines
fixes crash in moh when cachertclasses flag is used
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
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r269307 | rmudgett | 2010-06-09 18:54:38 +0200 (Wed, 09 Jun 2010) | 12 lines
Eliminate deadlock potential in dahdi_fixup().
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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r269308 | rmudgett | 2010-06-09 19:06:41 +0200 (Wed, 09 Jun 2010) | 2 lines
Add missing API function to sig_ss7: sig_ss7_fixup().
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r269346 | pabelanger | 2010-06-09 19:32:52 +0200 (Wed, 09 Jun 2010) | 19 lines
Merged revisions 269334 via svnmerge from
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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r269417 | russell | 2010-06-09 23:11:43 +0200 (Wed, 09 Jun 2010) | 6 lines
Resolve an invalid memory read on an event.
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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r269486 | qwell | 2010-06-09 23:38:33 +0200 (Wed, 09 Jun 2010) | 12 lines
Blocked revisions 269426 via svnmerge
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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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r269497 | russell | 2010-06-10 00:19:20 +0200 (Thu, 10 Jun 2010) | 9 lines
Merged revisions 269495 via svnmerge from
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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r269569 | russell | 2010-06-10 01:56:08 +0200 (Thu, 10 Jun 2010) | 2 lines
Attempt to fix FreeBSD build problem.
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r269602 | russell | 2010-06-10 02:32:31 +0200 (Thu, 10 Jun 2010) | 4 lines
Attempt to fix a FreeBSD build error by including sys/stat.h.
http://bamboo.asterisk.org/download/AST-TRUNKFREEBSD/build_logs/AST-TRUNKFREEBSD-187.log
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r269636 | tilghman | 2010-06-10 10:15:45 +0200 (Thu, 10 Jun 2010) | 16 lines
Merged revisions 269635 via svnmerge from
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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r269707 | kpfleming | 2010-06-10 14:28:17 +0200 (Thu, 10 Jun 2010) | 3 lines
Ensure that 'logger show channels' works properly when wildcards are used in logger.conf.
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r269711 | russell | 2010-06-10 15:17:51 +0200 (Thu, 10 Jun 2010) | 2 lines
Fix an off by one error that caused a unit test to occasionally crash.
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r269749 | mmichelson | 2010-06-10 19:14:38 +0200 (Thu, 10 Jun 2010) | 5 lines
Add documentation explaining PLC in Asterisk.
Review: https://reviewboard.asterisk.org/r/688/
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r269822 | mmichelson | 2010-06-10 21:34:03 +0200 (Thu, 10 Jun 2010) | 25 lines
Merged revisions 269821 via svnmerge from
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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r269889 | pabelanger | 2010-06-10 22:30:44 +0200 (Thu, 10 Jun 2010) | 8 lines
Remove ASTBINDIR variable
(closes issue #17031)
Reported by: pabelanger
Patches:
Makefile.ASTBINDIR.v2.patch uploaded by pabelanger (license 224)
Tested by: pabelanger, tilghman
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r269936 | tilghman | 2010-06-11 20:04:54 +0200 (Fri, 11 Jun 2010) | 2 lines
Remove lines from the output related to the backtrace itself.
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r269938 | tilghman | 2010-06-11 20:17:28 +0200 (Fri, 11 Jun 2010) | 7 lines
Add DBGetComplete event after a DBGetResponse.
(closes issue #16965)
Reported by: rrb3942
Patches:
DBGetComplete.patch uploaded by rrb3942 (license 1003)
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r269976 | tilghman | 2010-06-11 20:31:14 +0200 (Fri, 11 Jun 2010) | 15 lines
Merged revisions 269960 via svnmerge from
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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r270042 | pabelanger | 2010-06-11 22:14:13 +0200 (Fri, 11 Jun 2010) | 11 lines
Use pkg-config to find gmime libraries
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
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r270079 | pabelanger | 2010-06-12 20:55:47 +0200 (Sat, 12 Jun 2010) | 9 lines
Merged revisions 270078 via svnmerge from
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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r270151 | pabelanger | 2010-06-13 03:53:54 +0200 (Sun, 13 Jun 2010) | 3 lines
Reverting patch and reopening issue #16155, as patch breaks
FreeBSD / OSX builds.
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r270184 | tzafrir | 2010-06-13 11:16:25 +0200 (Sun, 13 Jun 2010) | 9 lines
bashism in configure script
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)
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r270219 | rmudgett | 2010-06-14 17:55:35 +0200 (Mon, 14 Jun 2010) | 6 lines
Add digit manipulation tag support to chan_dahdi/sig_pri like chan_misdn.
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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r270260 | qwell | 2010-06-14 21:41:43 +0200 (Mon, 14 Jun 2010) | 8 lines
Add option to get untruncated channel name from AGENT function.
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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r270298 | rmudgett | 2010-06-14 22:51:09 +0200 (Mon, 14 Jun 2010) | 5 lines
Extract sig_ss7_init_linkset() to sig_ss7.
Also found a place where sig_pri_init_pri() was inlined and called it
instead.
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r270332 | pabelanger | 2010-06-14 23:33:55 +0200 (Mon, 14 Jun 2010) | 21 lines
Merged revisions 270331 via svnmerge from
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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r270443 | lmadsen | 2010-06-15 14:51:37 +0200 (Tue, 15 Jun 2010) | 9 lines
Merged revisions 270442 via svnmerge from
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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r270519 | tilghman | 2010-06-15 19:06:23 +0200 (Tue, 15 Jun 2010) | 10 lines
Add distributed devicestate via the XMPP protocol.
(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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r270552 | tilghman | 2010-06-15 20:16:04 +0200 (Tue, 15 Jun 2010) | 2 lines
Argh, mixed declarations and code.
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r270584 | tilghman | 2010-06-15 20:26:26 +0200 (Tue, 15 Jun 2010) | 12 lines
Merged revisions 270583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270583 | tilghman | 2010-06-15 13:25:12 -0500 (Tue, 15 Jun 2010) | 5 lines
Variables have always been case-sensitive, so we should not be removing case-insensitive matches.
Bug reported via the -dev list. See
http://lists.digium.com/pipermail/asterisk-dev/2010-June/044510.html
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r270658 | twilson | 2010-06-15 22:18:04 +0200 (Tue, 15 Jun 2010) | 18 lines
Make contactdeny apply to src ip when nat=yes
chan_sip's "contactdeny" feature screens the "to be registered contact".
In case of nat=yes it should not use the address information from the
Contact header (which is not used at all for routing), but the source
IP address of the request.
Thus, if nat=yes and a client sends a request from a denied IP address
(e.g. by spoofing the src-IP address) it can bypass the screening.
This commit makes contactdeny apply to the src ip when nat=yes instead.
(closes issue #17276)
Reported by: klaus3000
Patches:
patch-asterisk-trunk-contactdeny.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
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r270660 | twilson | 2010-06-15 23:10:15 +0200 (Tue, 15 Jun 2010) | 15 lines
Don't send files twice and remove extra \r\n from header
After the manager http auth changes, we forgot to remove the manual
sending of the file. Also, ast_http_send adds two \r\n to the header that
is passed to it, so a trailing \r\n is removed from the Content-type
header. It might be better to change ast_http_send, but I don't like changing
the behavior of an API function.
(closes issue #17239)
Reported by: cjacobsen
Patches:
patch2.diff uploaded by cjacobsen (license 1029)
Tested by: lathama, cjacobsen
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r270692 | twilson | 2010-06-15 23:42:33 +0200 (Tue, 15 Jun 2010) | 9 lines
Don't continue sending the file when there has been an error
If there is a problem with a firmware file, Polycom phones will close the
connection. We were continuing to send the file anyway. There should be no
reason to continue sending a file if there is an error writing it.
(closes issue #16682)
Reported by: lmadsen
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r270726 | russell | 2010-06-16 00:48:12 +0200 (Wed, 16 Jun 2010) | 2 lines
Don't blow up if an ast_channel doesn't get allocated.
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r270801 | pabelanger | 2010-06-16 17:05:11 +0200 (Wed, 16 Jun 2010) | 9 lines
Update formatting for channelvariables.tex
(closes issue #17511)
Reported by: klaus3000
Patches:
channelvariables.tex-patch.txt uploaded by klaus3000 (license 65)
Tested by: pabelanger
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r270836 | jpeeler | 2010-06-16 18:45:07 +0200 (Wed, 16 Jun 2010) | 5 lines
Fix no call waiting caller ID
Clearing the callwaitcas flag in analog_call was causing the incoming D digit
to be ignored which triggers sending the caller ID.
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r270867 | dvossel | 2010-06-16 19:36:51 +0200 (Wed, 16 Jun 2010) | 28 lines
Merged revisions 270866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270866 | dvossel | 2010-06-16 12:35:29 -0500 (Wed, 16 Jun 2010) | 22 lines
fixes chan_iax2 race condition
There is code in chan_iax2.c that attempts to guarantee that only a single
active thread will handle a call number at a time. This code works once
the thread is added to an active_list of threads, but we are not currently
guaranteed that a newly activated thread will enter the active_list immediately
because it is left up to the thread to add itself after frames have been
queued to it. This means that if two frames come in for the same call number
at the same time, it is possible for them to grab two separate threads because
the first thread did not add itself to the active_list fast enough. This
causes some pretty complex problems.
This patch resolves this race condition by immediately adding an activated
thread to the active_list within the network thread and only depending on
the thread to remove itself once it is done processing the frames queued to
it. By doing this we are guaranteed that if another frame for the same call
number comes in at the same time, that this thread will immediately be found
in the active_list of threads.
Review: https://reviewboard.asterisk.org/r/720/
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r270936 | pabelanger | 2010-06-16 20:43:22 +0200 (Wed, 16 Jun 2010) | 8 lines
MSG_OOB flag on HANGUP packet removed.
Per Tilghman's request on IRC (#asterisk-bugs).
(closes issue #17506)
Reported by: brycebaril
Tested by: pabelanger, tilghman
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r270940 | dvossel | 2010-06-16 21:03:24 +0200 (Wed, 16 Jun 2010) | 9 lines
addition of G.719 pass-through support
(closes issue #16293)
Reported by: malcolmd
Patches:
g719.passthrough.patch.7 uploaded by malcolmd (license 924)
format_g719.c uploaded by malcolmd (license 924)
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r270974 | mnicholson | 2010-06-16 22:34:31 +0200 (Wed, 16 Jun 2010) | 11 lines
Set sin_family to AF_INET when doing lookups, also reset sin_port the first time the ip address changes.
(closes issue #17496)
Reported by: ManChicken
(closes issue #15827)
Reported by: DennisD
Patches:
dnsmgr_15827.patch uploaded by chappell (license 8)
Tested by: DennisD, gentlec, damage, wimpy
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r270981 | qwell | 2010-06-16 23:10:48 +0200 (Wed, 16 Jun 2010) | 11 lines
Merged revisions 270980 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines
Need to lock the agent chan before access its internal bits.
Pointed out by russellb on asterisk-dev mailing list.
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r270983 | qwell | 2010-06-16 23:12:25 +0200 (Wed, 16 Jun 2010) | 1 line
Fix the actual place that was pointed out, for previous commit.
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r270987 | pabelanger | 2010-06-16 23:17:39 +0200 (Wed, 16 Jun 2010) | 11 lines
Merged revisions 270979 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r270979 | pabelanger | 2010-06-16 17:10:05 -0400 (Wed, 16 Jun 2010) | 4 lines
Fixed typo in macro-page
Reported to #asterisk-dev by a student of jsmith.
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r271056 | dvossel | 2010-06-17 00:37:45 +0200 (Thu, 17 Jun 2010) | 2 lines
addition of more parse_uri test cases
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r271089 | pabelanger | 2010-06-17 02:30:51 +0200 (Thu, 17 Jun 2010) | 5 lines
option w[(secs)] incorrectly capitalized in xmldoc
(closes issue #17516)
Reported by: karlfife
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r271124 | mnicholson | 2010-06-17 17:11:55 +0200 (Thu, 17 Jun 2010) | 13 lines
Blocked revisions 271123 via svnmerge
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r271123 | mnicholson | 2010-06-17 10:11:27 -0500 (Thu, 17 Jun 2010) | 7 lines
Set sin_family in ast_get_ip_or_srv() and removed the 'last' member of the ast_dnsmgr_entry struct.
(closes issue #15827)
Reported by: DennisD
Patches:
(modified) dnsmgr_15827.patch uploaded by chappell (license 8)
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r271192 | jpeeler | 2010-06-17 17:34:08 +0200 (Thu, 17 Jun 2010) | 1 line
Change expected operation from error to debug message
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r271231 | dvossel | 2010-06-17 19:23:43 +0200 (Thu, 17 Jun 2010) | 9 lines
adds speex 16khz audio support
(closes issue #17501)
Reported by: fabled
Patches:
asterisk-trunk-speex-wideband-v2.patch uploaded by fabled (license 448)
Tested by: malcolmd, fabled, dvossel
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r271261 | dvossel | 2010-06-17 20:36:06 +0200 (Thu, 17 Jun 2010) | 9 lines
adds support for slin16 in sip
(closes issue #16153)
Reported by: kfister
Patches:
16153-1.6.2.0-rc5.patch uploaded by kfister (license 912)
slin16.sip.patch.1 uploaded by malcolmd (license 924)
Tested by: kfister, malcolmd
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r271262 | dvossel | 2010-06-17 20:45:32 +0200 (Thu, 17 Jun 2010) | 29 lines
retransmit response to BYE requests until timer J expires
According to RFC 3261 section 17.2.2, which describes non-INVITE server
transaction, when a dialog enters the Completed state it must destroy
the dialog after Timer J (T1*64) fires. For a BYE transaction Asterisk
terminates the dialog immediately during sip_hangup() when it should be
waiting T1*64 ms. This results in some odd behavior. For instance if
Asterisk receives a BYE and transmits a 200ok in response, if the endpoint
never receives the 200ok it will retransmit the BYE to which Asterisk
responds with a "481 Call leg/transaction does not exist" because the
dialog is already gone.
To resolve this I made a function called sip_scheddestroy_final(). This
differs slightly from sip_schedestroy() in that it enables a flag that
will prevent the destruction from ever being rescheduled or canceled
afterwards. It also prevents the pvt's needdestroy flag from being set
which triggers the destruction of the dialog within the do_monitor thread().
By using this function we are guaranteed destruction will not occur
until the scheduled time. This allows Asterisk to respond to any possible
retransmits for a dialog after we process the initial BYE request for T1*64 ms.
Other changes: I removed two instances where sip_cancel_destroy is used
right before calling sip_scheddestroy. sip_scheddestroy always calls
sip_cancel_destroy before scheduling the new destruction so it is completely
unnecessary.
Review: https://reviewboard.asterisk.org/r/694/
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r271300 | dvossel | 2010-06-17 23:23:41 +0200 (Thu, 17 Jun 2010) | 2 lines
fixes some coding guideline issue
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r271336 | jpeeler | 2010-06-18 20:36:55 +0200 (Fri, 18 Jun 2010) | 20 lines
Recorded merge of revisions 271335 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271335 | jpeeler | 2010-06-18 13:33:17 -0500 (Fri, 18 Jun 2010) | 13 lines
Eliminate deadlock potential in dahdi_fixup().
(This is a backport of 269307, committed to trunk by rmudgett.)
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.
(closes issue #17261)
Reported by: aragon
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r271341 | dvossel | 2010-06-18 20:59:05 +0200 (Fri, 18 Jun 2010) | 2 lines
file.c was truncating audio file formats to the lower 32bits.
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r271483 | jpeeler | 2010-06-18 23:32:09 +0200 (Fri, 18 Jun 2010) | 18 lines
Merged revisions 271399 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271399 | jpeeler | 2010-06-18 14:28:24 -0500 (Fri, 18 Jun 2010) | 11 lines
Fix crash when parsing some heavily nested statements in AEL on reload.
Due to the recursion used when compiling AEL in gen_prios, all the stack space
was being consumed when parsing some AEL that contained nesting 13 levels deep.
Changing a few large buffers to be heap allocated fixed the crash, although I
did not test how many more levels can now be safely used.
(closes issue #16053)
Reported by: diLLec
Tested by: jpeeler
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r271520 | tilghman | 2010-06-21 07:10:06 +0200 (Mon, 21 Jun 2010) | 8 lines
Add new application for declining counting words in multiple languages.
(closes issue #16869)
Reported by: chappell
Patches:
app_say_counted-20100317.c uploaded by chappell (license 8)
Tested by: chappell
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r271551 | dvossel | 2010-06-21 22:33:41 +0200 (Mon, 21 Jun 2010) | 2 lines
fixes logic error introduced by slin16 sip support
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r271553 | dvossel | 2010-06-21 22:46:22 +0200 (Mon, 21 Jun 2010) | 9 lines
fixes crash when From header URI is missing "sip:"
(closes issue #17437)
Reported by: klaus3000
Patches:
sip_crash uploaded by dvossel (license 671)
Tested by: klaus3000
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r271554 | jpeeler | 2010-06-21 22:46:53 +0200 (Mon, 21 Jun 2010) | 14 lines
Merged revisions 271552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271552 | jpeeler | 2010-06-21 15:37:47 -0500 (Mon, 21 Jun 2010) | 7 lines
Do not use sizeof to calculate size of a heap allocated character array.
Change left out from 271399.
(closes issue #16053)
Reported by: diLLec
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r271625 | dvossel | 2010-06-21 23:58:33 +0200 (Mon, 21 Jun 2010) | 7 lines
add speex 16khz sample frame so codec cost can be calculated
(closes issue #17534)
Reported by: fabled
Patches:
speex-wb-sample.diff uploaded by fabled (license 448)
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r271657 | tilghman | 2010-06-22 00:41:00 +0200 (Tue, 22 Jun 2010) | 2 lines
Conflict kqueue on OS X, since it doesn't work there yet, anyway.
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r271690 | mnicholson | 2010-06-22 14:58:28 +0200 (Tue, 22 Jun 2010) | 18 lines
Merged revisions 271689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271689 | mnicholson | 2010-06-22 07:52:27 -0500 (Tue, 22 Jun 2010) | 8 lines
Modify chan_sip's packet generation api to automatically calculate the Content-Length. This is done by storing packet content in a buffer until it is actually time to send the packet, at which time the size of the packet is calculated. This change was made to ensure that the Content-Length is always correct.
(closes issue #17326)
Reported by: kenner
Tested by: mnicholson, kenner
Review: https://reviewboard.asterisk.org/r/693/
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This change also adds an ast_str_copy_string() function (similar to ast_copy_string), that copies one ast_str into another, properly handling embedded nulls.
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r271762 | mnicholson | 2010-06-22 16:54:58 +0200 (Tue, 22 Jun 2010) | 15 lines
Merged revisions 271761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271761 | mnicholson | 2010-06-22 09:49:36 -0500 (Tue, 22 Jun 2010) | 9 lines
Allow users to specify a port for dundi peers.
(closes issue #17056)
Reported by: klaus3000
Patches:
dundi-peerport-patch-trunk.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000
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r271764 | mnicholson | 2010-06-22 17:08:39 +0200 (Tue, 22 Jun 2010) | 2 lines
Updated the CHANGES file documenting the addition of a configurable port in the dundi config file.
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r271831 | dvossel | 2010-06-22 17:46:22 +0200 (Tue, 22 Jun 2010) | 10 lines
fixes attended transfer behavior when both transferee and transferer hung up
If both the transferer and transferee of a attended transfer hangup before
the new channel picks up, the new channel should be hung up as well as it
has no endpoint to talk to. This mirrors the expected behavior used in 1.4.
(closes issue #17444)
Reported by: corruptor
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r271833 | russell | 2010-06-22 18:17:14 +0200 (Tue, 22 Jun 2010) | 5 lines
Change the method of retrieving the Asterisk version string.
Using this method makes it so res_fax doesn't have to be rebuilt on every
svn update.
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r271867 | russell | 2010-06-22 18:28:03 +0200 (Tue, 22 Jun 2010) | 7 lines
Resolve some errors that occur on a graceful shutdown.
Don't Finalize() if Initialize() did not succeed. This resulted in an error
about trying to Finalize() an invalid handle.
Also trim some trailing whitespace while in the area.
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r271868 | jpeeler | 2010-06-22 18:29:18 +0200 (Tue, 22 Jun 2010) | 14 lines
Add regular expression filtering for manager events.
This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.
(closes issue #14861)
Reported by: fnordian
Patches:
eventfilter3.patch uploaded by fnordian (license 110),
modified by me
Review: https://reviewboard.asterisk.org/r/673/
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r271903 | mnicholson | 2010-06-22 19:35:17 +0200 (Tue, 22 Jun 2010) | 15 lines
Merged revisions 271902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r271902 | mnicholson | 2010-06-22 12:31:57 -0500 (Tue, 22 Jun 2010) | 8 lines
Decrease the module ref count in sip_hangup when SIP_DEFER_BYE_ON_TRANSFER is set. This is necessary to keep the ref count correct.
(closes issue #16815)
Reported by: rain
Patches:
chan_sip-unref-fix.diff uploaded by rain (license 327) (modified)
Tested by: rain
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r271905 | dvossel | 2010-06-22 19:57:28 +0200 (Tue, 22 Jun 2010) | 6 lines
minor fixes for white/black event filters
This fixes a ref count leak in event filters and checks for
a filter container allocation failure during session creation.
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r271977 | dvossel | 2010-06-22 22:37:05 +0200 (Tue, 22 Jun 2010) | 11 lines
ignore CANCEL request after having already received final response to INVITE
RFC 3261 section 9 states that a CANCEL has no effect on a
request to a UAS that has already given a final response. This
patch checks to make sure there is a pending invite before
allowing a CANCEL request to be processed, otherwise it responds
to the CANCEL with a "481 Call/Transaction Does Not Exist".
Review: https://reviewboard.asterisk.org/r/697/
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r272014 | dvossel | 2010-06-23 00:11:50 +0200 (Wed, 23 Jun 2010) | 5 lines
fixes issue with 'dialplan remove extension blah' segfaulting with tab completion
(closes issue #17440)
Reported by: kobaz
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r272052 | russell | 2010-06-23 01:20:37 +0200 (Wed, 23 Jun 2010) | 6 lines
Don't try to lock/unlock an uninitialized lock on a dahdi_pri.
This small changes prevents destroy_all_channels() from accessing a lock on an
unused dahdi_pri struct, resolving a ton of ERRORs that get spewed out when
shutting Asterisk down gracefully.
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r272090 | mmichelson | 2010-06-23 19:08:34 +0200 (Wed, 23 Jun 2010) | 17 lines
Add extra protection for reinvite glare scenario.
Testing proved that if Asterisk sent a connected line reinvite, and
the endpoint to which the reinvite were being sent sent a reinvite, Asterisk
would not properly respond with a 491 response.
The reason is that on connected line reinvites, we set the dialog's invitestate
to INV_CALLING to prevent Asterisk from sending a rapid flurry of connected line
reinvites. For other reinvites we do not do this. Because of the current invitestate,
when Asterisk received the reinvite, we interpreted this as a spiraled INVITE, and thus
did not behave properly.
The fix for this is to not enter the loop detection or spiral logic in handle_request_invite
if the channel state is currently up. This way, no mid-call reinvites will be misinterpreted,
no matter what the nature of the reinvite may have been.
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r272109 | twilson | 2010-06-23 19:21:40 +0200 (Wed, 23 Jun 2010) | 12 lines
Make sure reload updates SLA config
Even if there are no stations or trunks defined, we need to start the sla
thread to make sure we get the reload event. Also, when doing a reload we need
to remove the existing trunks and stations or they end up hanging around.
(closes issue #16818)
Reported by: mbonin
Patches:
sla_reload.patch uploaded by twilson (license 396)
Tested by: twilson
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r272145 | tilghman | 2010-06-23 20:25:54 +0200 (Wed, 23 Jun 2010) | 8 lines
Load all lines from realtime, not just the first one.
(closes issue #17144)
Reported by: nahuelgreco
Patches:
20100513__issue17144__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r272146 | twilson | 2010-06-23 20:39:20 +0200 (Wed, 23 Jun 2010) | 2 lines
Don't start the sla thread unless we realy need it
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r272148 | tilghman | 2010-06-23 20:41:18 +0200 (Wed, 23 Jun 2010) | 12 lines
Recorded merge of revisions 272147 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r272147 | tilghman | 2010-06-23 13:40:28 -0500 (Wed, 23 Jun 2010) | 5 lines
Backport part of revision 136715 to fix callerid in voicemail text files (IMAP only).
(closes issue #16945)
Reported by: mneuhauser
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