[asterisk-commits] eliel: branch group/data_api_gsoc2009 r253532 - in /team/group/data_api_gsoc2...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Mar 19 11:12:19 CDT 2010
Author: eliel
Date: Fri Mar 19 11:11:52 2010
New Revision: 253532
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=253532
Log:
Merged revisions 234572,234631,234700,234776,234820,234855,234893,234897,234940,234976,235010,235053,235132,235226,235229,235265,235298,235342,235382,235422,235521,235573,235656,235660,235740,235774,235822,235904,235941,236027-236028,236063,236144,236183,236185-236186,236300,236304,236306,236308,236312,236358,236434,236510,236613,236667,236713,236756,236802,236847,236891,236893,236902,236982,237050,237098,237136,237213,237250,237284,237319,237323,237327,237406,237410,237414,237494,237574,237656,237699,237749,237802-237804,237839,237882,237920,237968,238010,238014,238091,238134,238181,238231,238313,238361,238405,238410,238412,238492,238527,238583,238630,238635,238716,238754,238795,238835,238916,239000,239037,239074,239111,239113-239114,239152,239191,239231,239245,239270,239308,239389,239427,239473,239520,239525,239571,239624-239625,239663,239665,239712,239719,239797,239834,239839,239920,239957,239996-239997,240039,240078,240129,240175,240179,240226,240271,240328-240329,240368,240411,240415,240419-240421,240499-240500,240505,240548,240552,240629,240667,240717,240769,240842,240887,240892,240969,240971,240973-240974,241012,241016,241097-241098,241143,241187-241188,241229-241230,241314-241315,241364,241366,241416,241503,241581,241627,241714,241766,241855-241856,241896,241938,242043,242092,242184-242186,242227,242317,242357,242424,242521,242607,242645,242719,242724,242729,242812,242857,242904,242919,242933,242954,242965,242967,242971,243075-243077,243118,243157-243158,243200,243242,243244,243266,243346,243391,243482,243487,243551,243571,243652,243693,243780,243860,243943,243986,244071,244154,244243,244331,244393,244443,244505,244547,244597-244598,244647,244688,244728-244729,244768-244769,244792,244887,244924,244927,244945,245006,245046,245090,245147,245192,245230,245268-245269,245306-245307,245344-245345,245382-245385,245423,245497,245578,245597,245610,245623-245624,245680,245727,245729,245793,245804,245864,245945,246022,246030,246070,246116,246200,246204,246208,246249,246260,246299,246338,246382,246420,246461,246546,246627,246667,246669,246710,246724,246789,246827,246863,246899,246942,246980-246981,246985,247035,247076,247124-247125,247169,247248,247262,247282,247295,247335,247380-247381,247423,247500,247503,247509,247609,247652,247736,247770,247787,247841,247914-247915,248003,248107-248108,248184,248225-248226,248269,248347,248397,248489,248533-248534,248584,248667,248671,248793,248861,248946,248952,249009-249013,249050,249056,249058,249101,249105,249186-249187,249235,249320,249405,249449,249491,249538,249623,249669-249670,249672,249759,249801,249889,249891-249893,249912,249925,249947,249950,250037,250042,250045,250051,250141,250190,250213,250235,250237,250246,250302-250303,250346,250392,250395,250478,250481,250565,250609,250614,250730,250787,250871,250913,250917,250979,251022,251038,251087,251137,251181,251221-251222,251262-251263,251310,251411,251475,251538,251585,251631,251677,251679-251680,251682,251736,251779,251821,251877,251881,251884,251946,251987,251989,252088-252089,252133,252229,252241,252277,252314,252362,252367,252442,252488,252534,252619,252623,252627,252709,252759-252760,252762,252767,252846,252848-252849,252976,252980,253004,253028,253032,253113,253205,253255-253256,253261,253345,253357,253378,253490 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r234572 | seanbright | 2009-12-14 11:08:09 -0500 (Mon, 14 Dec 2009) | 2 lines
The default rate for 'timing test' is actually 50/sec, not 100/sec as advertised.
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r234631 | lmadsen | 2009-12-14 12:19:58 -0500 (Mon, 14 Dec 2009) | 8 lines
Update IMAP build documentation.
Update the IMAP build documentation to show how to build on 64-bit
platforms.
(issue #16433)
Reported by: shrift
Tested by: lmadsen
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r234700 | tilghman | 2009-12-14 16:13:18 -0500 (Mon, 14 Dec 2009) | 12 lines
Merged revisions 234699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r234699 | tilghman | 2009-12-14 15:09:56 -0600 (Mon, 14 Dec 2009) | 5 lines
Deal with the situation where .flavor exists but .version does not.
Also make the script slightly more portable, in keeping with autoconf syntax.
(closes issue #14737)
Reported by: davidw
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r234776 | qwell | 2009-12-14 16:32:03 -0500 (Mon, 14 Dec 2009) | 12 lines
Allow tonelist as argument to ReadExten.
ReadExten already supported playing a tonezone from indications.conf.
It now has the ability to use a tonelist like 440+480/2000|0/4000
(closes issue #15185)
Reported by: jcovert
Patches:
app_readexten.c.patch uploaded by jcovert (license 551)
Tested by: qwell
Patch modified by me, to maintain backwards compatibility.
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r234820 | tilghman | 2009-12-14 18:16:00 -0500 (Mon, 14 Dec 2009) | 7 lines
Allow greetings-only mailboxes for Voicemail.
(closes issue #15132)
Reported by: floletarmo
Patches:
voicemail_changes.patch uploaded by floletarmo (license 784)
(with some additional changes by me)
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r234855 | alecdavis | 2009-12-14 19:54:44 -0500 (Mon, 14 Dec 2009) | 9 lines
ast_stream_and_wait(chan,dir-usingkeypad) didn't capture the dialled DTMF.
(closes issue #16409)
Reported by: alecdavis
Tested by: alecdavis
Patch
bug_16409.diff.txt uploaded by alecdavis (license 585)
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r234893 | alecdavis | 2009-12-14 21:29:50 -0500 (Mon, 14 Dec 2009) | 9 lines
fixes escape to extensions 'o' and 'a', for digits '0' and '*'
(closes issue #16437)
Reported by: alecdavis
Tested by: alecdavis
Patch
extension_o_a_fix.diff.txt uploaded by alecdavis (license 585)
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r234897 | alecdavis | 2009-12-14 22:04:59 -0500 (Mon, 14 Dec 2009) | 9 lines
restarts busydetector (if enabled) when DTMF is received.
(closes issue #16389)
Reported by: alecdavis
Tested by: alecdavis
Patch
dtmf_busydetector.diff.txt uploaded by alecdavis (license 585)
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r234940 | alecdavis | 2009-12-14 22:26:49 -0500 (Mon, 14 Dec 2009) | 2 lines
Whitespace.
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r234976 | alecdavis | 2009-12-15 02:18:31 -0500 (Tue, 15 Dec 2009) | 7 lines
Support option 'n', as applications like Playback, Background etc.
Suggested on asterisk-dev as trivial application change.
Reported by: alecdavis
Tested by: alecdavis
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r235010 | kpfleming | 2009-12-15 09:35:46 -0500 (Tue, 15 Dec 2009) | 5 lines
spandsp does in fact support V.17 modulation at 14.4 kilobits per second,
so we should generate T38MaxBitRate of 14400 (even though that doesn't really
affect the FAX transmission much at all)
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r235053 | tilghman | 2009-12-15 10:33:49 -0500 (Tue, 15 Dec 2009) | 11 lines
Merged revisions 235052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235052 | tilghman | 2009-12-15 09:29:24 -0600 (Tue, 15 Dec 2009) | 4 lines
Mandatory argument checking
(closes issue #16446)
Reported by: nicchap
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r235132 | dvossel | 2009-12-15 13:43:06 -0500 (Tue, 15 Dec 2009) | 14 lines
reverse minor sip registration regression
A registration regression caused by a code tweak in (issue #14331)
and a bug fix in (issue #15539) caused some sip registration
config entries to be constructed incorrectly. Origially
issue #14331 contained the code tweak as well as a bug fix, but since
the issue was reported as a tweak the bug fix portion was moved into
issue #15539. Both the tweak and the bug fix contained minor incorrect
logic that resulted in some SIP registrations to fail.
(issue #14331)
(issue #15539)
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r235226 | jpeeler | 2009-12-15 18:41:20 -0500 (Tue, 15 Dec 2009) | 7 lines
Change match criteria existence in ast_channel_cmp_cb to use ast_strlen_zero.
(closes issue #16161)
Reported by: may213
Patches:
core-show-channel.patch uploaded by may213 (license 454)
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r235229 | tilghman | 2009-12-15 18:51:05 -0500 (Tue, 15 Dec 2009) | 2 lines
Is it Friday yet?
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r235265 | jpeeler | 2009-12-15 19:31:53 -0500 (Tue, 15 Dec 2009) | 12 lines
Enhance AMI redirect to allow channels to be redirected to different places.
New parameters ExtraContext, ExtraExtension, and ExtraPriority have been added
to redirect the second channel to a different location. Previously, it was only
possible to redirect both channels to the same place.
(closes issue #15853)
Reported by: haakon
Patches:
trunk-manager.c.patch uploaded by haakon (license 880)
Tested by: jpeeler
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r235298 | jsmith | 2009-12-16 00:24:58 -0500 (Wed, 16 Dec 2009) | 11 lines
Merged revisions 235181 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235181 | jsmith | 2009-12-15 15:07:55 -0600 (Tue, 15 Dec 2009) | 4 lines
Add a line showing that we can use CIDR notation.
patch by jsmith, after discussion with jtodd
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r235342 | jpeeler | 2009-12-16 15:25:27 -0500 (Wed, 16 Dec 2009) | 8 lines
Add auth_policy option to jabber.conf for auto user registration.
The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
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r235382 | jpeeler | 2009-12-16 19:52:03 -0500 (Wed, 16 Dec 2009) | 5 lines
Fix call forwarding for analog phones.
(closes issue #16440)
Reported by: mmichelson
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r235422 | tilghman | 2009-12-17 12:19:08 -0500 (Thu, 17 Dec 2009) | 15 lines
Merged revisions 235421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235421 | tilghman | 2009-12-17 11:17:51 -0600 (Thu, 17 Dec 2009) | 8 lines
Use context from which Macro is executed, not macro context, if applicable.
Also, ensure that the extension COULD match, not just that it won't match more.
(closes issue #16113)
Reported by: OrNix
Patches:
20091216__issue16113.diff.txt uploaded by tilghman (license 14)
Tested by: OrNix
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r235521 | file | 2009-12-17 18:21:07 -0500 (Thu, 17 Dec 2009) | 3 lines
Remove some old code for going to the 'fax' extension when a T.38 switchover occurs. This would have
already happened when we detected the CNG tone so this was basically a noop.
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r235573 | tilghman | 2009-12-18 16:19:43 -0500 (Fri, 18 Dec 2009) | 9 lines
Merged revisions 235572 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) | 2 lines
Point to the typical missing package, not the cryptic "termcap support".
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r235656 | tilghman | 2009-12-18 17:40:46 -0500 (Fri, 18 Dec 2009) | 9 lines
Merged revisions 235652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235652 | tilghman | 2009-12-18 16:39:30 -0600 (Fri, 18 Dec 2009) | 2 lines
Revise verbiage, per #asterisk-dev discussion
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r235660 | jpeeler | 2009-12-18 17:51:37 -0500 (Fri, 18 Dec 2009) | 55 lines
Merged revisions 235635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines
Correct CDR dispositions for BUSY/FAILED
This patch is simple in that it reorders the disposition defines so that the fix
for issue 12946 works properly (the default CDR disposition was changed to
AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to
ensure all CDR records are written.
The side effects of CDR changes are scary, so I'm documenting the test cases
performed to attempt to catch any regressions. The following tests were all
performed using 1.4 rev 195881 vs head (235571) + patch:
A calls B
C calls B (busy)
Hangup C
Hangup A
(Both SIP and features)
A calls B
A blind transfers to C
Hangup C
(Both SIP and features)
A calls B
A attended transfers to C
Hangup C
A calls B
A attended transfers to C (SIP)
C blind transfers to A (features)
Hangup A
All of the test scenario CDRs matched.
The following tests were performed just with the patch to ensure proper operation
(with unanswered=yes):
exten =>s,1,Answer
exten =>s,n,ResetCDR(w)
exten =>s,n,ResetCDR(w)
exten =>s,1,ResetCDR(w)
exten =>s,n,ResetCDR(w)
(closes issue #16180)
Reported by: aatef
Patches:
bug16180.patch uploaded by jpeeler (license 325)
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r235740 | alecdavis | 2009-12-19 03:59:31 -0500 (Sat, 19 Dec 2009) | 9 lines
app_dial optional parameter to option 'r' to allow play indication from indications.conf
(closes issue #14504)
Reported by: alecdavis
Tested by: alecdavis,jsmith
Patch
app_dial.play_ring_indications.diff7.txt uploaded by alecdavis (license 585)
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r235774 | alecdavis | 2009-12-20 03:22:35 -0500 (Sun, 20 Dec 2009) | 8 lines
restarts busydetector (if enabled) when DTMF is received after call is bridged.
(closes issue 0016389)
Reported by: alecdavis
Tested by: alecdavis
Patch
dtmf_busydetector.diff2.txt uploaded by alecdavis (license 585)
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r235822 | tilghman | 2009-12-21 12:00:46 -0500 (Mon, 21 Dec 2009) | 15 lines
Merged revisions 235821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235821 | tilghman | 2009-12-21 10:45:03 -0600 (Mon, 21 Dec 2009) | 8 lines
Send parking lot announcement to the channel which parked the call, not the park-ee.
(closes issue #16234)
Reported by: yeshuawatso
Patches:
20091210__issue16234.diff.txt uploaded by tilghman (license 14)
20091221__issue16234__1.4.diff.txt uploaded by tilghman (license 14)
Tested by: yeshuawatso
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r235904 | kpfleming | 2009-12-21 13:51:17 -0500 (Mon, 21 Dec 2009) | 1 line
Change all refererences to 1.6.3 to be 1.8, since that will be the next feature release
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r235941 | jpeeler | 2009-12-21 14:54:20 -0500 (Mon, 21 Dec 2009) | 20 lines
Merged revisions 235940 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r235940 | jpeeler | 2009-12-21 13:43:41 -0600 (Mon, 21 Dec 2009) | 13 lines
Change Monitor to not assume file to write to does not contain pathing.
227944 changed the fname_base argument to always append the configured monitor
path. This change was necessary to properly compare files for uniqueness.
If a full path is given though, nothing needs to be appended and that is
handled correctly now.
(closes issue #16377)
(closes issue #16376)
Reported by: bcnit
Patches:
res_monitor.c-issue16376-1.patch uploaded by dant (license 670)
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r236027 | dvossel | 2009-12-22 11:09:11 -0500 (Tue, 22 Dec 2009) | 20 lines
Unit Test Framework API
The Unit Test Framework is a new API that manages registration and
execution of unit tests in Asterisk with the purpose of verifying the
operation of C functions. The Framework consists of a single test
manager accompanied by a list of registered test functions defined
within the code. A test is defined, registered, and unregistered
from the framework using a set of macros which allow the test code
to only be compiled within asterisk when the TEST_FRAMEWORK flag is
enabled in menuselect. This allows the test code to exist in the
same file as the C functions it intends to verify. Registered tests
may be viewed and executed via a set of new CLI commands. CLI commands
are also present for generating and exporting test results into xml
and txt formats.
For more information and use cases please refer to the documentation
provided at the beginning of the test.h file.
Review: https://reviewboard.asterisk.org/r/447/
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r236028 | dvossel | 2009-12-22 11:11:47 -0500 (Tue, 22 Dec 2009) | 2 lines
update CHANGES to reflect the addition of the test framework
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r236063 | dvossel | 2009-12-22 12:00:08 -0500 (Tue, 22 Dec 2009) | 18 lines
Merged revisions 236062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236062 | dvossel | 2009-12-22 10:58:19 -0600 (Tue, 22 Dec 2009) | 11 lines
fixes issue with p->method incorrectly set to ACK
It is possible for a second ACK to come in for a retransmitted message.
If an ack does not match an unacked message in our queue, restore the previous
p->method as this ACK is completely ignored.
(closes issue #16295)
Reported by: omolenkamp
Patches:
issue16295_v2.diff uploaded by dvossel (license 671)
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r236144 | dvossel | 2009-12-22 16:53:46 -0500 (Tue, 22 Dec 2009) | 7 lines
fixes iax "can't compress subclass 4294967295" error
(closes issue #16456)
Reported by: dvossel
Tested by: dvossel
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r236183 | tilghman | 2009-12-22 21:52:30 -0500 (Tue, 22 Dec 2009) | 2 lines
Actually use tmp for something (brings trunk back into sync with 1.6 branches).
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r236185 | tilghman | 2009-12-22 22:03:47 -0500 (Tue, 22 Dec 2009) | 2 lines
Allow test_heap.c to compile when AST_DEVMODE is true, but TEST_FRAMEWORK is false
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r236186 | tilghman | 2009-12-22 22:07:48 -0500 (Tue, 22 Dec 2009) | 11 lines
Merged revisions 236184 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) | 4 lines
If EXEC only gets a single argument, don't crash when the second is used.
(closes issue #16504)
Reported by: bklang
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r236300 | tilghman | 2009-12-23 13:25:27 -0500 (Wed, 23 Dec 2009) | 7 lines
AGI may be invoked from outside the dialplan
(closes issue #16510)
Reported by: atis
Patches:
20091223__issue16510.diff.txt uploaded by tilghman (license 14)
Tested by: atis
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r236304 | dvossel | 2009-12-23 13:39:37 -0500 (Wed, 23 Dec 2009) | 12 lines
new parameter 'R' to the Queue application
The 'R' argument stops moh and indicates ringing once the agent is
ringing. This allows the person in the queue to know their call
is potentially about to be answered.
(closes issue #16384)
Reported by: haakon
Patches:
new_app_queue.c.patch uploaded by haakon (license 880)
Tested by: haakon, loloski, dvossel
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r236306 | dvossel | 2009-12-23 13:45:54 -0500 (Wed, 23 Dec 2009) | 4 lines
update CHANGES to reflect new 'R' app_queue option plus a minor optimization to the feature patch
(issue #16384)
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r236308 | dvossel | 2009-12-23 14:14:05 -0500 (Wed, 23 Dec 2009) | 17 lines
QUEUE_MEMBER(..., ready) counts only ready agents, not free agents wrapping up
The QUEUE_MEMBER dialplan function can return total members,
logged-in members and "free" members count. A member is counted
as "free" immediately after his call ends, even though its wrap-up
time, if specified in queues.conf, has not yet expired, and the
queue will not actually route a call to it.
This Patch introduces a new "ready" option that only counts
free agents no longer in the wrap up time period.
(closes issue #16240)
Reported by: kkm
Patches:
appqueue-memberfun-readyoption-trunk.diff uploaded by kkm (license 888)
Tested by: kkm, dvossel
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r236312 | dvossel | 2009-12-23 14:17:42 -0500 (Wed, 23 Dec 2009) | 2 lines
Update CHANGES to reflect new QUEUE_MEMBER option, "ready"
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r236358 | kpfleming | 2009-12-26 10:27:44 -0500 (Sat, 26 Dec 2009) | 9 lines
Merged revisions 236357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236357 | kpfleming | 2009-12-26 09:26:17 -0600 (Sat, 26 Dec 2009) | 1 line
update to latest releases with zero uid/gid
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r236434 | tilghman | 2009-12-27 13:20:53 -0500 (Sun, 27 Dec 2009) | 9 lines
Merged revisions 236433 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236433 | tilghman | 2009-12-27 12:19:38 -0600 (Sun, 27 Dec 2009) | 2 lines
Turn on colors in the daemon, since there's many requests for it on Ubuntu.
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r236510 | seanbright | 2009-12-28 07:44:58 -0500 (Mon, 28 Dec 2009) | 19 lines
Merged revisions 236509 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines
Avoid a crash with large numbers of MeetMe conferences.
Similar to changes made to Queue(), when we have large numbers of conferences in
meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and
crash, so instead just use a single fixed buffer.
(closes issue #16509)
Reported by: Kashif Raza
Patches:
20091223_16509.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec 2009) | 14 lines
Merged revisions 236585 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) | 7 lines
Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces.
There was conditional code (based on build platform) to optioinally wrap
PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions
of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add
a configure-time check for it.
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r236667 | tilghman | 2009-12-28 12:37:46 -0500 (Mon, 28 Dec 2009) | 4 lines
Use recommended option, not deprecated option.
(closes issue #16515)
Reported by: ManChicken
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r236713 | qwell | 2009-12-28 17:09:40 -0500 (Mon, 28 Dec 2009) | 8 lines
Allow "REMAINDER" to function properly in expressions.
(closes issue #16427)
Reported by: wdoekes
Patches:
ast16-reminder-remainder.patch uploaded by wdoekes (license 717)
Tested by: wdoekes
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r236756 | transnexus | 2009-12-29 05:59:55 -0500 (Tue, 29 Dec 2009) | 4 lines
1. Updated for OSP Toolkit 3.6.0.
2. Added service type ported number query.
3. Formated code.
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r236802 | tilghman | 2009-12-29 18:05:45 -0500 (Tue, 29 Dec 2009) | 7 lines
Shut down the SIP session timers more gracefully, in order to prevent a possible crash.
(closes issue #16452)
Reported by: corruptor
Patches:
20091221__issue16452.diff.txt uploaded by tilghman (license 14)
Tested by: corruptor
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r236847 | tilghman | 2009-12-30 12:53:29 -0500 (Wed, 30 Dec 2009) | 4 lines
When the field is blank, don't warn about the field being unable to be coerced, just skip the column.
(closes http://lists.digium.com/pipermail/asterisk-dev/2009-December/041362.html)
Reported by Nic Colledge on the -dev list, fixed by me.
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r236891 | jpeeler | 2009-12-30 15:25:58 -0500 (Wed, 30 Dec 2009) | 8 lines
Blocked revisions 236890 via svnmerge
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r236890 | jpeeler | 2009-12-30 14:25:11 -0600 (Wed, 30 Dec 2009) | 2 lines
Remove conflicting function definitions (asterisk.h) so LOW_MEMORY compiles.
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r236893 | jpeeler | 2009-12-30 15:34:41 -0500 (Wed, 30 Dec 2009) | 11 lines
Fix compiling with LOW_MEMORY.
Modified handle_verbose to be LOW_MEMORY aware, removed old RTP related code
in chan_sip.
(closes issue #16381)
Reported by: michael_iedema
Patches:
ast_complete_source_filename.patch uploaded by michael iedema (license 942)
modified by me
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r236902 | jpeeler | 2009-12-30 16:09:28 -0500 (Wed, 30 Dec 2009) | 2 lines
One more LOW_MEMORY compile fix.
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r236982 | tilghman | 2009-12-30 16:59:18 -0500 (Wed, 30 Dec 2009) | 16 lines
Merged revisions 236981 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r236981 | tilghman | 2009-12-30 15:57:10 -0600 (Wed, 30 Dec 2009) | 9 lines
Don't queue frames to channels that have no means to process them.
(closes issue #15609)
Reported by: aragon
Patches:
20091230__issue16521__1.4__chan_local_only.diff.txt uploaded by tilghman (license 14)
Tested by: aragon
Review: https://reviewboard.asterisk.org/r/452/
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r237050 | qwell | 2009-12-30 17:30:21 -0500 (Wed, 30 Dec 2009) | 17 lines
Add app_voicemail and say.c support for Vietnamese.
Also add an XXX comment that I'm baffled nobody has ever complained about. We
say "first message", and then we go into language-specific stuff where we
proceed to say..."first message".
(closes issue #15053)
Reported by: dinhtrung
Patches:
vietnamese.ods uploaded by dinhtrung (license 776)
app_voicemail.c.diff uploaded by dinhtrung (license 776)
(closes issue #15626)
Reported by: dinhtrung
Patches:
say.c.diff uploaded by dinhtrung (license 776)
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r237098 | may | 2009-12-30 18:51:32 -0500 (Wed, 30 Dec 2009) | 11 lines
small q931 processing and signalling corrections
don't decode UUIE from Q931StatusMessage
clean call without callIdentifier data
don't start tcs/msd exchange procedure after call proceeding received
(closes issue #16365)
Reported by: benngard2
Tested by: may213, benngard2
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r237136 | oej | 2010-01-02 04:54:22 -0500 (Sat, 02 Jan 2010) | 10 lines
Merged revisions 237135 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r237135 | oej | 2010-01-02 10:52:30 +0100 (Lör, 02 Jan 2010) | 2 lines
Release memory of the contact acl before unloading module
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r237213 | tilghman | 2010-01-02 11:35:35 -0500 (Sat, 02 Jan 2010) | 2 lines
global_contact_ha was renamed in trunk
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r237250 | transnexus | 2010-01-03 22:38:29 -0500 (Sun, 03 Jan 2010) | 3 lines
1. Added reporting operator names in AuthReq.
2. Added retrieving operator names from AuthRsp and exporting them.
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r237284 | oej | 2010-01-04 02:55:30 -0500 (Mon, 04 Jan 2010) | 4 lines
- Disable res_pktccops by default
- Add dependency in chan_mgcp that was missing
- Add a small amount of doc to the source code
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r237319 | tilghman | 2010-01-04 11:20:03 -0500 (Mon, 04 Jan 2010) | 10 lines
Merged revisions 237318 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r237318 | tilghman | 2010-01-04 10:18:59 -0600 (Mon, 04 Jan 2010) | 3 lines
It's also possible for the Local channel to directly execute an Application.
Reviewboard: https://reviewboard.asterisk.org/r/452/
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r237323 | jpeeler | 2010-01-04 11:24:51 -0500 (Mon, 04 Jan 2010) | 5 lines
Fix timeout for AGI command speech recognize.
(closes issue #16297)
Reported by: semond
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r237327 | dvossel | 2010-01-04 11:39:11 -0500 (Mon, 04 Jan 2010) | 10 lines
app_queue segfaults if realtime field uniqueid is NULL
(closes issue #16385)
Reported by: haakon
Patches:
app_queue.c.patch uploaded by haakon (license 880)
app_queue.c.patch_v2 uploaded by dvossel (license 671)
Tested by: haakon
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r237406 | tilghman | 2010-01-04 13:28:28 -0500 (Mon, 04 Jan 2010) | 23 lines
Merged revisions 237405 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines
Add a flag to disable the Background behavior, for AGI users.
This is in a section of code that relates to two other issues, namely
issue #14011 and issue #14940), one of which was the behavior of
Background when called with a context argument that matched the current
context. This fix broke FreePBX, however, in a post-Dial situation.
Needless to say, this is an extremely difficult collision of several
different issues. While the use of an exception flag is ugly, fixing all
of the issues linked is rather difficult (although if someone would like
to propose a better solution, we're happy to entertain that suggestion).
(closes issue #16434)
Reported by: rickead2000
Patches:
20091217__issue16434.diff.txt uploaded by tilghman (license 14)
20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: rickead2000
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r237410 | tilghman | 2010-01-04 13:42:10 -0500 (Mon, 04 Jan 2010) | 7 lines
Further reduce the encoded blank values back to blank in the realtime API.
(closes issue #16533)
Reported by: sergee
Patches:
200100104__issue16533.diff.txt uploaded by tilghman (license 14)
Tested by: sergee
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r237414 | tilghman | 2010-01-04 14:03:20 -0500 (Mon, 04 Jan 2010) | 2 lines
Oops, didn't compile (thanks, kpfleming)
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r237494 | tilghman | 2010-01-04 15:59:01 -0500 (Mon, 04 Jan 2010) | 15 lines
Merged revisions 237493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r237493 | tilghman | 2010-01-04 14:57:35 -0600 (Mon, 04 Jan 2010) | 8 lines
Regression in issue #15421 - Pattern matching
(closes issue #16482)
Reported by: wdoekes
Patches:
astsvn-16482-betterfix.diff uploaded by wdoekes (license 717)
20091223__issue16482.diff.txt uploaded by tilghman (license 14)
Tested by: wdoekes, tilghman
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r237574 | tilghman | 2010-01-04 16:48:20 -0500 (Mon, 04 Jan 2010) | 13 lines
Merged revisions 237573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r237573 | tilghman | 2010-01-04 15:45:46 -0600 (Mon, 04 Jan 2010) | 6 lines
Bounds checking for input string
(closes issue #16407)
Reported by: qwell
Patches:
20100104__issue16407.diff.txt uploaded by tilghman (license 14)
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r237656 | mvanbaak | 2010-01-05 11:08:12 -0500 (Tue, 05 Jan 2010) | 6 lines
Make CLI command 'mixmonitor start|stop <channel> work again.
(closes issue #16534)
Reported by: jlaguilar
Fix as suggested by jlaguilar in the bugreport
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r237699 | russell | 2010-01-05 12:16:01 -0500 (Tue, 05 Jan 2010) | 14 lines
Merged revisions 237697 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r237697 | russell | 2010-01-05 11:13:28 -0600 (Tue, 05 Jan 2010) | 7 lines
Change a NOTICE log message to DEBUG where it belongs.
(closes issue #16479)
Reported by: alexrecarey
(closes SWP-577)
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r237749 | russell | 2010-01-05 12:26:50 -0500 (Tue, 05 Jan 2010) | 2 lines
Fix build of utility apps that include utils.c.
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r237802 | mmichelson | 2010-01-05 13:42:36 -0500 (Tue, 05 Jan 2010) | 3 lines
Spell "aficionado" like someone who isn't stupid.
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r237803 | mmichelson | 2010-01-05 13:46:19 -0500 (Tue, 05 Jan 2010) | 9 lines
Add a missing part of the connected line work into trunk.
Part of the work done for connected line was to add an optional
argument to the 'f' option to allow for the connected party information
of the outgoing channel to be set to the argument provided. This was
overlooked during the merge of the work to trunk and is being added
back now. The CHANGES file has also been updated to note this change.
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r237804 | rmudgett | 2010-01-05 13:57:29 -0500 (Tue, 05 Jan 2010) | 1 line
Removed unused parameters from analog_available() and sig_pri_available().
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r237839 | dvossel | 2010-01-05 14:29:47 -0500 (Tue, 05 Jan 2010) | 19 lines
fixes subscriptions being lost after 'module reload'
During a module reload if multiple extension configs are present,
such as both extensions.conf and extensions.ael, watchers for one
config's hints will be lost during the merging of the other config.
This happens because hint watchers are only preserved for the
current config being merged. The old context list is destroyed
after the merging takes place, meaning any watchers that were not
perserved will be removed.
Now all hints are preserved during merging regardless of what config
file is being merged. These hints are only restored if they
are present within the new context list.
(closes issue #16093)
Reported by: jlaroff
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r237882 | mmichelson | 2010-01-05 15:56:50 -0500 (Tue, 05 Jan 2010) | 3 lines
Mismerged a bit.
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r237920 | dvossel | 2010-01-05 18:08:50 -0500 (Tue, 05 Jan 2010) | 16 lines
fixes holdtime playback issue in app_queue
When reporting hold time, the number of seconds should be mod 60.
Otherwise audio playback could be something like "2 minutes 123 seconds"
rather than "2 minutes 3 seconds".
Also, the "minute" sound file is missing, so for the moment until
that file can be created the "minutes" file is used instead.
(closes issue #16168)
Reported by: nickilo
Patches:
patch-unified-trunk-rev-222176 uploaded by nickilo (license )
Tested by: nickilo, wonderg
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r237968 | tilghman | 2010-01-06 01:53:23 -0500 (Wed, 06 Jan 2010) | 2 lines
Whoa, duplicate setting (dead code).
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r238010 | russell | 2010-01-06 10:19:10 -0500 (Wed, 06 Jan 2010) | 14 lines
Merged revisions 238009 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r238009 | russell | 2010-01-06 09:18:22 -0600 (Wed, 06 Jan 2010) | 7 lines
Resolve a crash due to an ast_frame not being fully initialized.
(closes issue #16531)
Reported by: john8675309
(closes SWP-615)
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r238014 | seanbright | 2010-01-06 10:35:43 -0500 (Wed, 06 Jan 2010) | 16 lines
Fix reading samples from format_mp3 after ast_seekstream/ast_tellstream.
There is a bug when using ast_seekstream/ast_tellstream with format_mp3 in that
the file read position is not reset before attempting to read samples. So when
we seek to determine the maximum size of the file (as in res_agi's STREAM FILE)
we weren't then resetting the file pointer so that we could properly read
samples. This patch addresses that (in a similar manner to format_wav.c).
(closes issue #15224)
Reported by: rbd
Patches:
20091230_addons_1.4_issue15224.diff uploaded by seanbright (license 71)
Tested by: rbd, seanbright
Review: https://reviewboard.asterisk.org/r/453
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r238091 | dvossel | 2010-01-06 11:36:02 -0500 (Wed, 06 Jan 2010) | 11 lines
fixes test.c compile issue when TEST_FRAMEWORK is not enabled
The ast_test_status_update() function is defined in test.h.
When TEST_FRAMEWORK is not enabled a macro is defined as a no-op
place holder for this function. The macro did not contain
the correct number of arguments. This caused a compile error.
Much thanks to wdoekes for reporting the issue and supplying the
patch!
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r238134 | jpeeler | 2010-01-06 14:05:06 -0500 (Wed, 06 Jan 2010) | 10 lines
Fix channel name comparison for bridge application.
The channel name comparison was not comparing the whole string and therefore
if one channel name was a substring of the other, the bridge would fail.
(closes issue #16528)
Reported by: telecos82
Patches:
res_features_r236843.diff uploaded by telecos82 (license 687)
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r238181 | jpeeler | 2010-01-06 15:37:18 -0500 (Wed, 06 Jan 2010) | 8 lines
Fix misreverting from 177158.
(closes issue #15725)
Reported by: shanermn
Patches:
v1-15725.patch uploaded by dimas (license 88)
Tested by: shanermn
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r238231 | tilghman | 2010-01-06 16:45:17 -0500 (Wed, 06 Jan 2010) | 11 lines
Merged revisions 238230 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r238230 | tilghman | 2010-01-06 15:41:55 -0600 (Wed, 06 Jan 2010) | 4 lines
Revise documentation on disposition values to the actual values used.
(closes issue #16289)
Reported by: wdoekes
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r238313 | tzafrir | 2010-01-07 04:14:57 -0500 (Thu, 07 Jan 2010) | 2 lines
Document the usefulness of explicit udp:// in the register string
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r238361 | dvossel | 2010-01-07 13:58:23 -0500 (Thu, 07 Jan 2010) | 8 lines
cli 'queue show' formatting fix. queue name was truncated over 12 characters
(closes issue #16078)
Reported by: RoadKill
Patches:
quequename_limit.patch uploaded by ppyy (license 906)
Tested by: dvossel
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r238405 | dvossel | 2010-01-07 15:00:31 -0500 (Thu, 07 Jan 2010) | 8 lines
Change in sip show channels display format allowing more digits for CID
(closes issue #16459)
Reported by: Rzadzins
Patches:
chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)
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r238410 | dvossel | 2010-01-07 15:06:43 -0500 (Thu, 07 Jan 2010) | 13 lines
Blocked revisions 238409 via svnmerge
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r238409 | dvossel | 2010-01-07 14:06:00 -0600 (Thu, 07 Jan 2010) | 7 lines
Change in sip show channels display format allowing more digits for CID
(closes issue 0016459)
Reported by: Rzadzins
Patches:
chan_sip_longer_cid.patch uploaded by Rzadzins (license 953)
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r238412 | dvossel | 2010-01-07 15:15:27 -0500 (Thu, 07 Jan 2010) | 16 lines
Merged revisions 238411 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r238411 | dvossel | 2010-01-07 14:14:25 -0600 (Thu, 07 Jan 2010) | 10 lines
fixes crash in "scheduled_destroy" in chan_iax
A signed short was used to represent a callnumber. This is makes
it possible to attempt to access the iaxs array with a negative
index.
(closes issue #16565)
Reported by: jensvb
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r238492 | dvossel | 2010-01-07 15:42:27 -0500 (Thu, 07 Jan 2010) | 14 lines
fixes ast_transfer stall until hangup if called with a channel that doesn't support transfers
ast_transfer sets res to 0 if there is no technology transfer function,
but then tests for it to be negative before deciding to do an early exit.
As a result, it will will wait for an AST_CONTROL_TRANSFER message that
will never come.
(closes issue #16424)
Reported by: davidw
Patches:
Issue_16424_trunk_234134.patch uploaded by davidw (license 780)
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