Russell Bryant [Sun, 7 Feb 2010 21:56:04 +0000 (21:56 +0000)]
Add an MD5 test module.
Review: https://reviewboard.asterisk.org/r/491/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245307
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Russell Bryant [Sun, 7 Feb 2010 21:43:11 +0000 (21:43 +0000)]
Fix a couple of spelling errors, and add format module dependencies.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245306
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Russell Bryant [Sun, 7 Feb 2010 20:57:42 +0000 (20:57 +0000)]
Tweak formatting and add minor updates to some comments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245269
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Russell Bryant [Sun, 7 Feb 2010 20:56:57 +0000 (20:56 +0000)]
Remove an extra space.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245268
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Mark Michelson [Sun, 7 Feb 2010 19:51:54 +0000 (19:51 +0000)]
Remove parsing of constantssrc from reload_config.
This config option is already handled by the function handle_common_options
and it is unnecessary to parse the value again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245230
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Mark Michelson [Sat, 6 Feb 2010 14:43:03 +0000 (14:43 +0000)]
Remove useless sip options related to hash table size.
First off, these options weren't actually doing anything.
By the time the options were parsed, the peer and dialog
containers had already been allocated with their default
values.
Second, hash table size is something that doesn't really
make sense to change in a config file. If a user is that
interested in changing the hashtable size, he can modify
the source itself.
I have removed the parsing of the hash_peer, hash_user,
and hash_dialog options. I have removed the hash_user_size
variable altogether since it is not used at all. I also
changed hash_peer_size and hash_dialog_size to be constant,
and have changed the symbols to be in all caps as constants
typically are. I have also removed the entire section in
sip.conf.sample regarding configurable hashtable sizes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245192
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David Vossel [Fri, 5 Feb 2010 21:21:05 +0000 (21:21 +0000)]
fixes astobj2 unlinking of multiple objects when OBJ_MULTIPLE was disabled
When OBJ_MULTIPLE was off but OBJ_UNLINK was on, all the items in a bucket
were being unlinked instead of just the first match. This fixes that.
Review: https://reviewboard.asterisk.org/r/490/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245147
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Jeff Peeler [Fri, 5 Feb 2010 19:26:22 +0000 (19:26 +0000)]
Merged revisions 245044 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r245044 | kpfleming | 2010-02-05 12:32:29 -0600 (Fri, 05 Feb 2010) | 5 lines
Remove contrib/firmware directory as it is empty
Remove explicit license for IAXy firmware as it is no longer included in the tree
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245090
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Tilghman Lesher [Fri, 5 Feb 2010 19:07:18 +0000 (19:07 +0000)]
Merge tests that verify the same thing. (Oops.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245046
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David Vossel [Fri, 5 Feb 2010 18:12:41 +0000 (18:12 +0000)]
adds total call numbers available to 'iax2 show callnumber usage' cli output
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245006
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Terry Wilson [Fri, 5 Feb 2010 17:20:24 +0000 (17:20 +0000)]
Fix crash on 32-bit for users not using https
(closes issue #16778)
Reported by: pitel
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: twilson, pitel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244945
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Sean Bright [Fri, 5 Feb 2010 17:05:32 +0000 (17:05 +0000)]
Merged revisions 244926 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r244926 | seanbright | 2010-02-05 12:03:35 -0500 (Fri, 05 Feb 2010) | 1 line
Update main copyright date.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244927
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David Vossel [Fri, 5 Feb 2010 16:59:06 +0000 (16:59 +0000)]
fixes issue with sip registry not having correct default expiry
default expiry was not being set correctly for a registry object.
Thanks to ebroad for reporting the issue and testing the patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244924
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David Vossel [Fri, 5 Feb 2010 04:22:15 +0000 (04:22 +0000)]
fixes memory leak in astobj2 test
ao2_iterator_destroy was not being used on the
iterator during the test. This resulted in the
container never actually being destroyed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244887
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Jeff Peeler [Thu, 4 Feb 2010 23:21:49 +0000 (23:21 +0000)]
Blocked revisions 244785 via svnmerge
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r244785 | jpeeler | 2010-02-04 17:20:21 -0600 (Thu, 04 Feb 2010) | 22 lines
Change channel state on local channels for busy,answer,ring.
Previously local channels channel state never changed. This became problematic
when the state of the other side of the local channel was lost, for example
during a masquerade. Changing the state of the local channel allows for the
scenario to be detected when the channel state is set to ringing, but the peer
isn't ringing. The specific problem scenario is described in 164201. Although
this was noted on one of the issues, here is the tested dialplan verified to
work:
exten => 9700,1,Dial(Local/*9700@default&Local/#9700@default)
exten => *9700,1,Set(GLOBAL(TESTCHAN)=${CHANNEL:0:${MATH(${LEN(${CHANNEL})}-1):0:2}}1)
exten => *9700,n,wait(3) ;3 works, 1 did not
exten => *9700,n,Dial(SIP/5001)
exten => #9700,1,Wait(1) ;1 works, 3 did not
exten => #9700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1)
(closes issue #14992)
Reported by: davidw
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244792
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David Vossel [Thu, 4 Feb 2010 23:16:30 +0000 (23:16 +0000)]
parse_moved_contact tries to parse contact_name twice
parse_moved_contact attempts to remove a quoted string
twice, and the first try wasn't even being done correctly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244769
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Tilghman Lesher [Thu, 4 Feb 2010 22:43:33 +0000 (22:43 +0000)]
Try to make ast_format_str_reduce fail...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244768
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Tilghman Lesher [Thu, 4 Feb 2010 18:47:21 +0000 (18:47 +0000)]
Oops
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244729
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Tilghman Lesher [Thu, 4 Feb 2010 18:46:12 +0000 (18:46 +0000)]
Define a small set of constant return values
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244728
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David Vossel [Thu, 4 Feb 2010 15:36:33 +0000 (15:36 +0000)]
fix truncated format string in 'test show registered'
When using the 'test show registered' cli command the 'Test Results'
category was truncating the last few characters making it look like
'Test Resul'. I also expanded other parts of the format to better
represent how long function names and categories will likely be.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244688
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Richard Mudgett [Thu, 4 Feb 2010 00:12:31 +0000 (00:12 +0000)]
Add ignore *.i files property to the new channels/sip directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244647
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Jeff Peeler [Wed, 3 Feb 2010 20:48:36 +0000 (20:48 +0000)]
Add some additional option support for non-default parking lots.
The options are: parkedcallparking, parkedcallhangup, parkedcallrecording, and
parkedcalltransfers. Previously these options were only available for the
default parking lot.
(closes issue #16641)
Reported by: bluecrow76
Patches:
asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76 (license 270)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244598
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David Vossel [Wed, 3 Feb 2010 20:33:32 +0000 (20:33 +0000)]
-----Changes -----
New files
- channels/sip/sip.h – A new header for shared #define, enum, and struct
definitions.
- channels/sip/include/sip_utils.h – sip util functions shared among
the all the sip APIs
- channels/sip/include/config_parser.h – sip config-parser API
- channels/sip/config_parser.c – Contains sip.conf parsing helper functions
with unit tests.
- channels/sip/include/reqresp_parser.h – sip request response parser API
- channels/sip/reqresp_parser.c – Contains sip request and response parsing
helper functions with unit tests.
New Unit Tests
- sip_parse_uri_test
- sip_parse_host_test
- sip_parse_register_line_test
Code Refactoring
- All reusable #define, enum, and struct definitions were moved out of chan_sip.c
into sip.h. During this process formatting changes were made to comments
in both sip.h and chan_sip.c in order to better adhere to the coding guidelines.
- The beginnings of three new sip APIs, sip-utils.h, config-parser.h,
reqresp-parser.h using existing chan_sip.c functions.
- parse_uri() and get_calleridname() were moved from chan_sip.c to request-parser.c
along with unit tests for both functions.
- sip_parse_host() and sip_parse_register_line() were moved from chan_sip.c to
config-parser.c along with unit tests for both functions.
Changes to parse_uri()
-removal of the options parameter. It was never used and did not behave correctly.
-additional check for [?header] field. When this field was present, the transport
type was not being set correctly.
----- Overview -----
This patch is introduced with the hope that unit tests for all our sip parsing
functions will be written soon. chan_sip is a huge file, and with the addition of
each unit test chan_sip is going to grow larger and harder to maintain. I'm proposing
we begin refactoring chan_sip, starting with the parsing functions. With each parsing
function we move into a separate helper file, a unit test should accompany it. I've
attempted to lay down the ground work for this change by creating two new parser
helper files (config-parser.c and reqresp-parser.c) and moving all shared structs,
enums, and defines from chan_sip.c into a shared sip.h file. We can't verify everything
in Asterisk using unit tests, but string parsing is one area where unit tests make
the most sense. By beginning to restructure the code in this way, chan_sip not only
becomes less bloated, but Asterisk as a whole will become more stable.
Review: https://reviewboard.asterisk.org/r/477/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244597
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Mark Michelson [Wed, 3 Feb 2010 19:26:53 +0000 (19:26 +0000)]
Initialize counters in ast_sched_report so that resulting data is not bogus.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244547
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Tilghman Lesher [Wed, 3 Feb 2010 18:34:29 +0000 (18:34 +0000)]
The chanvar= setting should inherit the entire list of variables, not just the first one.
(closes issue #16359)
Reported by: raarts
Patches:
dahdi-setvars.diff uploaded by raarts (license 937)
Tested by: raarts
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244505
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David Vossel [Tue, 2 Feb 2010 22:27:23 +0000 (22:27 +0000)]
fixes crash during T.38 negotiation caused by invalid or missing FaxMaxDatagram field
AST-2010-001
(closes issue #16634)
Reported by: krn
(closes issue #16724)
Reported by: barthpbx
(closes issue #16517)
Reported by: bklang
(closes issue #16485)
Reported by: elsto
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244443
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Tilghman Lesher [Tue, 2 Feb 2010 20:32:29 +0000 (20:32 +0000)]
Properly respect GOSUB_RESULT as to what to do with the master channel.
Previously, we would parse GOSUB_RESULT, but not actually do anything with it.
Also, allow GOSUB_RETVAL to be inherited back across a peer/master channel.
(closes issue #16687)
Reported by: bklang
Patches:
app_dial-preserve-gosub_retval.patch uploaded by bklang (license 919)
(with modifications)
(closes issue #16686)
Reported by: bklang
Patches:
app_dial-respect-gosub_result.patch uploaded by bklang (license 919)
(with modifications)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244393
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Tilghman Lesher [Tue, 2 Feb 2010 18:54:33 +0000 (18:54 +0000)]
Correct some off-by-one errors, especially when expressions don't contain expected spaces.
Also include the tests provided by the reporter, as regression tests.
(closes issue #16667)
Reported by: wdoekes
Patches:
astsvn-func_match-off-by-one.diff uploaded by wdoekes (license 717)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244331
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Tilghman Lesher [Mon, 1 Feb 2010 23:16:12 +0000 (23:16 +0000)]
Merged revisions 244242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r244242 | tilghman | 2010-02-01 17:13:44 -0600 (Mon, 01 Feb 2010) | 11 lines
Backup and restore original textfile, for prosthesis (gerund of prepend).
Also, fix menuselect such that changing voicemail build options correctly
causes rebuild.
(closes issue #16415)
Reported by: tomo1657
Patches:
prepention.patch uploaded by tomo1657 (license 484)
(with modifications by me to backport to 1.4)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244243
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Tilghman Lesher [Mon, 1 Feb 2010 18:39:32 +0000 (18:39 +0000)]
Blocked revisions 244151 via svnmerge (fixed a different way in trunk and 1.6.x)
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r244151 | tilghman | 2010-02-01 12:38:37 -0600 (Mon, 01 Feb 2010) | 8 lines
When a transferer hangs up during an attended transfer BEFORE the transfer is answered, don't stop playing MOH.
(closes issue #16513)
Reported by: litnimax
Patches:
atxfer_moh_16513.patch uploaded by gknispel proformatique (license 261)
Tested by: litnimax
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244154
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Tilghman Lesher [Mon, 1 Feb 2010 17:53:39 +0000 (17:53 +0000)]
Merged revisions 244070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r244070 | tilghman | 2010-02-01 11:46:31 -0600 (Mon, 01 Feb 2010) | 16 lines
Revert previous chan_local fix (r236981) and fix instead by destroying expired frames in the queue.
(closes issue #16525)
Reported by: kobaz
Patches:
20100126__issue16525.diff.txt uploaded by tilghman (license 14)
20100129__issue16525__1.6.0.diff.txt uploaded by tilghman (license 14)
Tested by: kobaz, atis
(closes issue #16581)
Reported by: ZX81
(closes issue #16681)
Reported by: alexr1
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244071
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Jeff Peeler [Thu, 28 Jan 2010 22:37:15 +0000 (22:37 +0000)]
Optimization to manager events.
When potentially sending manager events, return immediately if there are no
sessions or hooks. Also, avoid locking the hooks list if it is empty.
(issue #16455)
Reported by: atis
Patches:
manager_hooks_trunk.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243986
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Tilghman Lesher [Thu, 28 Jan 2010 20:00:09 +0000 (20:00 +0000)]
Informational message, not an error.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243943
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Russell Bryant [Thu, 28 Jan 2010 18:35:15 +0000 (18:35 +0000)]
Add a missing line terminator for T.38 SDP.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243860
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Russell Bryant [Thu, 28 Jan 2010 15:07:23 +0000 (15:07 +0000)]
Merged revisions 243779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r243779 | russell | 2010-01-28 09:03:17 -0600 (Thu, 28 Jan 2010) | 2 lines
Fix a bogus third argument to ast_copy_string().
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Jeff Peeler [Wed, 27 Jan 2010 20:37:33 +0000 (20:37 +0000)]
Merged revisions 243691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010) | 5 lines
Revert 243570, I should have looked at this closer. Will reopen the issue, but
am leaving the review closed as the change was pointless.
(issue #16488)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243693
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Jeff Peeler [Wed, 27 Jan 2010 20:06:08 +0000 (20:06 +0000)]
expand code based appreviation of AST_CONFIG_DIR to configuration directory
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243652
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Jeff Peeler [Wed, 27 Jan 2010 18:49:52 +0000 (18:49 +0000)]
Merged revisions 243570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010) | 9 lines
Extend announcement URL used with Queue from 80 chars to PATH_MAX.
(closes issue #16488)
Reported by: syspert
Patches:
soundfilelen.pacth-2 uploaded by syspert (license 938)
Review: https://reviewboard.asterisk.org/r/475/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243571
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Jeff Peeler [Wed, 27 Jan 2010 18:29:49 +0000 (18:29 +0000)]
Add new option to asterisk.conf (lockconfdir) to protect conf dir during reloads
(closes issue #16358)
Reported by: raarts
Patches:
lockconfdir.diff uploaded by raarts (license 937)
modified by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243551
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Mark Michelson [Wed, 27 Jan 2010 18:08:02 +0000 (18:08 +0000)]
Merged revisions 243486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r243486 | mmichelson | 2010-01-27 12:06:43 -0600 (Wed, 27 Jan 2010) | 3 lines
Use a safe list traversal while checking for duplicate vars in pbx_builtin_setvar_helper.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243487
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Russell Bryant [Wed, 27 Jan 2010 17:32:07 +0000 (17:32 +0000)]
Fix the ability to specify an OSP token for an outbound IAX2 call.
When this patch was originally submitted, the code allowed for the token to be
set via a channel variable. I decided that a cleaner approach would be to
integrate it into the CHANNEL() function. Unfortunately, that is not a suitable
approach. It's not possible to get the value set on the channel soon enough
using that method. So, go back to the simple channel variable method.
(closes issue #16711)
Reported by: homesick
Patches:
iax-svn.diff uploaded by homesick (license 91)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243482
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David Vossel [Tue, 26 Jan 2010 23:56:57 +0000 (23:56 +0000)]
Merged revisions 243390 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r243390 | dvossel | 2010-01-26 17:55:49 -0600 (Tue, 26 Jan 2010) | 9 lines
fixes bug with channel receiving wrong privileges after call parking
(closes issue #16429)
Reported by: Yasuhiro Konishi
Patches:
features.c.diff uploaded by Yasuhiro Konishi (license 947)
Tested by: dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243391
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David Ruggles [Tue, 26 Jan 2010 20:49:57 +0000 (20:49 +0000)]
Code clean up in app_senddtmf
Pushes code clean up done in app_externalivr back
into app_senddtmf
Review: https://reviewboard.asterisk.org/r/473/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243346
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Jeff Peeler [Tue, 26 Jan 2010 18:20:55 +0000 (18:20 +0000)]
Merged revisions 243258 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r243258 | jpeeler | 2010-01-26 12:19:10 -0600 (Tue, 26 Jan 2010) | 2 lines
Remove unnecessary code in ast_read as issue 16058 has been fully solved now.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243266
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Jeff Peeler [Tue, 26 Jan 2010 18:07:57 +0000 (18:07 +0000)]
Fix crash resulting from frames with invalid data pointers.
In ast_frdup the frame data union does not get set to point to malloced memory
if the datalen is zero, so make sure to handle the same case in ast_frisolate
appropriately.
(closes issue #16058)
Reported by: atis
Patches:
bug16058-fix.patch uploaded by jpeeler (license 325)
Tested by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243244
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David Vossel [Tue, 26 Jan 2010 17:40:02 +0000 (17:40 +0000)]
modify 'test show registered' cli output format
In order to improve readability, the output from 'test show
registered' has been modified to truncate fields to fit within
the format output if they are over a certain length.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243242
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David Vossel [Tue, 26 Jan 2010 16:30:08 +0000 (16:30 +0000)]
RFC compliant uri and display-name encode/decode
1. URI Encoding
This patch changes ast_uri_encode()'s behavior when doreserved is enabled.
Previously when doreserved was enabled only a small set of reserved
characters were encoded. This set was comprised primarily of the reserved
characters defined in RFC3261 section 25.1, but contained other characters as
well. Rather than only escaping the reserved set, doreserved now escapes
all characters not within the unreserved set as defined by RFC 3261 and
RFC 2396. Also, the 'doreserved' variable has been renamed to 'do_special_char'
in attempts to avoid confusion.
When doreserve is not enabled, the previous logic of only encoding the
characters <= 0X1F and > 0X7f remains, except for the '%' character, which
must always be encoded as it signifies a HEX escaped character during the decode
process.
2. URI Decoding: Break up URI before decode.
In chan_sip.c ast_uri_decode is called on the entire URI instead of it's
individual parts after it is parsed. This is not good as ast_uri_decode
can introduce special characters back into the URI which can mess up parsing.
This patch resolves this by not decoding a URI until parsing is completely
done. There are many instances where we check to see if pedantic checking
is enabled before we decode a URI. In these cases a new macro,
SIP_PEDANTIC_DECODE, is used on the individual parsed segments of the URI
rather than constantly putting if (pedantic) { decode() } checks everywhere
in the code. In the areas where ast_uri_decode is not dependent upon
pedantic checking this macro is not used, but decoding is still moved to
each individual part of the URI. The only behavior that should change from
this patch is the time at which decoding occurs.
Since I had to look over every place URI parsing occurs to create this
patch, I found several places where we use duplicate code for parsing.
To consolidate the code, those areas have updated to use the parse_uri()
function where possible.
3. SIP display-name decoding according to RFC3261 section 25.
To properly decode the display-name portion of a FROM header, chan_sip's
get_calleridname() function required a complete re-write. More information
about this change can be found in the comments at the beginning of this function.
4. Unit Tests.
Unit tests for ast_uri_encode, ast_uri_decode, and get_calleridname() have been
written. This involved the addition of the test_utils.c file for testing the
utils api.
(closes issue #16299)
Reported by: wdoekes
Patches:
astsvn-16299-get_calleridname.diff uploaded by wdoekes (license 717)
get_calleridname_rewrite.diff uploaded by dvossel (license 671)
Tested by: wdoekes, dvossel, Nick_Lewis
Review: https://reviewboard.asterisk.org/r/469/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243200
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Russell Bryant [Tue, 26 Jan 2010 15:46:53 +0000 (15:46 +0000)]
Log the variable name being tested.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243158
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Russell Bryant [Tue, 26 Jan 2010 15:35:40 +0000 (15:35 +0000)]
Update test_substitution to show failures in the test log.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243157
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Russell Bryant [Tue, 26 Jan 2010 15:16:59 +0000 (15:16 +0000)]
Update func_aes to its pre-ast_str_substitution state.
This change makes the AES tests in test_substitution.c pass. We still need to
work through what's going wrong in the ast_str version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243118
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Tilghman Lesher [Tue, 26 Jan 2010 01:56:24 +0000 (01:56 +0000)]
Fixing last errors in the conversion, though it appears that the AES_* functions are still broken.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243077
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Tilghman Lesher [Tue, 26 Jan 2010 01:41:47 +0000 (01:41 +0000)]
Using a dummy channel causes CDR() testing to fail.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243076
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Tilghman Lesher [Tue, 26 Jan 2010 01:35:19 +0000 (01:35 +0000)]
Wish I had gotten to the review before this got submitted, because there's failures we need to address.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243075
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Tilghman Lesher [Mon, 25 Jan 2010 21:51:41 +0000 (21:51 +0000)]
Merged revisions 242969 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242969 | tilghman | 2010-01-25 15:50:22 -0600 (Mon, 25 Jan 2010) | 2 lines
Err, and use the new menuselect define, too.
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Tilghman Lesher [Mon, 25 Jan 2010 21:38:33 +0000 (21:38 +0000)]
Merged revisions 242966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242966 | tilghman | 2010-01-25 15:36:33 -0600 (Mon, 25 Jan 2010) | 2 lines
Only rebuild parsers by an option in menuselect
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Russell Bryant [Mon, 25 Jan 2010 21:32:38 +0000 (21:32 +0000)]
Make unit test modules depend on TEST_FRAMEWORK instead of off by default.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242965
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Russell Bryant [Mon, 25 Jan 2010 21:25:23 +0000 (21:25 +0000)]
Convert test_substitution module to the unit test API.
Review: https://reviewboard.asterisk.org/r/474/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242954
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Alexandr Anikin [Mon, 25 Jan 2010 21:20:02 +0000 (21:20 +0000)]
small corrections in call clearing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242933
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Olle Johansson [Mon, 25 Jan 2010 21:13:20 +0000 (21:13 +0000)]
Change api for pbx_builtin_setvar to actually return error code if a function can't be written to.
This patch removes code that was duplicated from pbx.c to manager.c
in order to prevent API change in released versions of Asterisk.
There are propably also other places that would benefit from reading the
return code and react if a function returns error codes on writing a value into it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242919
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Olle Johansson [Mon, 25 Jan 2010 20:27:59 +0000 (20:27 +0000)]
Merged revisions 242850 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242850 | oej | 2010-01-25 21:03:38 +0100 (Mån, 25 Jan 2010) | 2 lines
Report error when writing to functions returns error in AMI setvar action
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Tilghman Lesher [Mon, 25 Jan 2010 20:18:15 +0000 (20:18 +0000)]
Merged revisions 242852 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242852 | tilghman | 2010-01-25 14:15:45 -0600 (Mon, 25 Jan 2010) | 2 lines
Restore FreeBSD to able-to-compile-ish-mode
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Terry Wilson [Mon, 25 Jan 2010 18:01:08 +0000 (18:01 +0000)]
Fix INTERNAL_OBJ error on stop when calendars.conf missing
Initialize the calendars container before calling load_config and return FAILURE
on allocation failure. Also, use the AST_MODULE_LOAD_* values for return values.
Thanks to rmudgett for pointing out the error and the need to use the defined
values for return
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242812
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Tilghman Lesher [Mon, 25 Jan 2010 05:45:00 +0000 (05:45 +0000)]
Merged revisions 242728 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242728 | tilghman | 2010-01-24 23:42:22 -0600 (Sun, 24 Jan 2010) | 2 lines
Buildbot pointed out an error (thanks, buildbot!)
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Tilghman Lesher [Mon, 25 Jan 2010 05:34:33 +0000 (05:34 +0000)]
Merged revisions 242723 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242723 | tilghman | 2010-01-24 23:33:37 -0600 (Sun, 24 Jan 2010) | 2 lines
Oops, should have used CMD_PREFIX, not ECHO_PREFIX, for the commands.
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Tilghman Lesher [Mon, 25 Jan 2010 05:30:33 +0000 (05:30 +0000)]
Merged revisions 242683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242683 | tilghman | 2010-01-24 23:13:28 -0600 (Sun, 24 Jan 2010) | 2 lines
Make the build of the Asterisk expression parser match that of the AEL parser.
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Alexandr Anikin [Sun, 24 Jan 2010 22:42:11 +0000 (22:42 +0000)]
AST_CONTROL_CONNECTED_LINE frame type processing added to setup DisplayIE field
incorrect q.931 message order filtered on incoming calls (first msg must be setup,
next must be not setup)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242645
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Sean Bright [Sun, 24 Jan 2010 21:49:17 +0000 (21:49 +0000)]
Instead of crashing, allocate our header ast_str before we try to use it.
(closes issue #16680)
Reported by: lmadsen
Patches:
issue16680_20100122.patch uploaded by seanbright (license 71)
Tested by: lmadsen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242607
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Tilghman Lesher [Sun, 24 Jan 2010 06:40:31 +0000 (06:40 +0000)]
Merged revisions 242520 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
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Tilghman Lesher [Fri, 22 Jan 2010 21:45:18 +0000 (21:45 +0000)]
Merged revisions 242423 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242423 | tilghman | 2010-01-22 15:44:18 -0600 (Fri, 22 Jan 2010) | 7 lines
Rebuild from flex, bison sources when necessary.
(issue #14629)
Reported by: Marquis
Patches:
20100121__issue14629.diff.txt uploaded by tilghman (license 14)
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David Ruggles [Fri, 22 Jan 2010 16:20:43 +0000 (16:20 +0000)]
Add send DTMF feature to ExternalIVR app
Implemented a new command 'D' that allows client
IVRs to send DTMF digits to the channel.
(closes issue #16615)
Reported by: thedavidfactor
Review: https://reviewboard.asterisk.org/r/465/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242357
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Tilghman Lesher [Fri, 22 Jan 2010 15:09:12 +0000 (15:09 +0000)]
The irony of not compile-testing a test program before committing is killing me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242317
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Olle Johansson [Fri, 22 Jan 2010 09:28:34 +0000 (09:28 +0000)]
Merged revisions 242226 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242226 | oej | 2010-01-22 10:19:30 +0100 (Fre, 22 Jan 2010) | 3 lines
Initialize notify_types to NULL
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Russell Bryant [Fri, 22 Jan 2010 04:57:03 +0000 (04:57 +0000)]
Update the doxygenification of some comments.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242186
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Russell Bryant [Fri, 22 Jan 2010 04:50:38 +0000 (04:50 +0000)]
Convert scheduler API entry order test to the test API.
Review: https://reviewboard.asterisk.org/r/470/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242185
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Russell Bryant [Fri, 22 Jan 2010 04:49:26 +0000 (04:49 +0000)]
Add test API usage example to test_skel.c.
Review: https://reviewboard.asterisk.org/r/471/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242184
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Mark Michelson [Thu, 21 Jan 2010 22:37:55 +0000 (22:37 +0000)]
Add missing argument to ast_calloc calls.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242092
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Olle Johansson [Thu, 21 Jan 2010 21:05:56 +0000 (21:05 +0000)]
Make sure we initialize the ast_ha structure with ast_calloc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242043
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Sean Bright [Thu, 21 Jan 2010 15:27:42 +0000 (15:27 +0000)]
Merged revisions 241932 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r241932 | seanbright | 2010-01-21 10:25:46 -0500 (Thu, 21 Jan 2010) | 5 lines
Fix configure check for PTHREAD_ONCE_INIT when manually adding -Wall to CFLAGS.
(closes issue #16666)
Reported by: romain_proformatique
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Tilghman Lesher [Thu, 21 Jan 2010 15:14:55 +0000 (15:14 +0000)]
Formats are inconsistent between even 32-bit and 64-bit Linux. Use casts to ensure both compile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241896
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Russell Bryant [Thu, 21 Jan 2010 14:10:51 +0000 (14:10 +0000)]
Point to a useful reference on the XML output format.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241856
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Russell Bryant [Thu, 21 Jan 2010 13:59:41 +0000 (13:59 +0000)]
Modify test results XML format to match the JUnit format.
When this code was developed, we came up with our own XML format for the test
output. I have since started looking at integration with other tools, namely
continuous integration frameworks, and this format seems to be supported
across a number of applications. With these changes in place, I was able
to get Atlassian Bamboo to interpret the test results.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241855
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Tilghman Lesher [Thu, 21 Jan 2010 05:54:30 +0000 (05:54 +0000)]
Merged revisions 241765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r241765 | tilghman | 2010-01-20 23:53:17 -0600 (Wed, 20 Jan 2010) | 2 lines
Guard against division by zero.
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David Vossel [Wed, 20 Jan 2010 21:14:47 +0000 (21:14 +0000)]
rtp timestamp to timeval calculation fix
The rtp timestamp to timeval calculation was only
accurate for 8kHz audio. This patch corrects this.
Review: https://reviewboard.asterisk.org/r/468/
SWP-648
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241714
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David Vossel [Wed, 20 Jan 2010 20:01:07 +0000 (20:01 +0000)]
Merged revisions 241626 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r241626 | dvossel | 2010-01-20 14:00:04 -0600 (Wed, 20 Jan 2010) | 6 lines
fixes parsing error in Makefile.
Some echo lines were missing "; . Thanks to jparker for pointing out
the problem.
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Alec L Davis [Wed, 20 Jan 2010 17:49:30 +0000 (17:49 +0000)]
Add Calling and Called Subaddress to CDR record
Requires 'callingsubaddr' and 'calledsubaddr' fields in backend cdr.
(closes issue #16600)
Reported by: alecdavis
Patches:
cdr_subaddr.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
Review: https://reviewboard.asterisk.org/r/460/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241581
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Kevin P. Fleming [Wed, 20 Jan 2010 13:01:00 +0000 (13:01 +0000)]
Fix up compile breakage from ast_tvdiff_ms() API change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241503
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Alec L Davis [Wed, 20 Jan 2010 08:18:45 +0000 (08:18 +0000)]
Update CDR variables as pbx starts
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call,
by executing ast_cdr_update() early in __ast_pbx run.
Reverts sig_pri changes in trunk that are specific to isdn technology only.
(closes issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241416
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Jeff Peeler [Tue, 19 Jan 2010 22:59:53 +0000 (22:59 +0000)]
Initialize data on the stack so that Park doesn't interpret random arguments.
passdata was only being set in pbx_substitue_variables when arguments were
passed.
(closes issue #16406)
(closes issue #16586)
Reported by: DLNoah
Patches:
bug16586v2.patch uploaded by jpeeler (license 325)
Tested by: DLNoah
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241366
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Tilghman Lesher [Tue, 19 Jan 2010 22:41:36 +0000 (22:41 +0000)]
Enable SendText to send strings in encoded format.
See http://lists.digium.com/pipermail/asterisk-users/2010-January/243462.html
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241364
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Jeff Peeler [Tue, 19 Jan 2010 18:51:19 +0000 (18:51 +0000)]
small correction from 241314
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241315
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Jeff Peeler [Tue, 19 Jan 2010 18:46:11 +0000 (18:46 +0000)]
Merged revisions 241227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19 Jan 2010) | 13 lines
Fix deadlock in agent_read by removing call to agent_logoff.
One must always lock the agents list lock before the agent private. agent_read
locks the private immediately, so locking the agents list lock is not an
option (which is what agent_logoff requires). Because agent_read already
has access to the agent private all that is necessary is to do the required
hanging up that agent_logoff performed.
(closes issue #16321)
Reported by: valon24
Patches:
bug16321.patch uploaded by jpeeler (license 325)
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Jason Parker [Tue, 19 Jan 2010 17:42:10 +0000 (17:42 +0000)]
Allow parallel make (-j) to work properly.
After some back and forth with the reporter, we came up with the necessary changes.
(closes issue #16489)
Reported by: Chainsaw
Patches:
asterisk-1.6.2.1-parallel-make-minimal.patch uploaded by Chainsaw (license 723)
Tested by: Chainsaw, qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241230
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Jason Parker [Tue, 19 Jan 2010 17:41:19 +0000 (17:41 +0000)]
Blocked revisions 241228 via svnmerge
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r241228 | qwell | 2010-01-19 11:41:00 -0600 (Tue, 19 Jan 2010) | 8 lines
Allow parallel make (-j) to work properly.
1.4 changes are quite different from the others.
(issue #16489)
Reported by: Chainsaw
Tested by: qwell
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Tilghman Lesher [Tue, 19 Jan 2010 00:28:49 +0000 (00:28 +0000)]
Create iterative method for querying SRV results, and use that for finding AGI servers.
(closes issue #14775)
Reported by: _brent_
Patches:
20091215__issue14775.diff.txt uploaded by tilghman (license 14)
hagi-5.patch uploaded by brent (license 388)
Tested by: _brent_
Reviewboard: https://reviewboard.asterisk.org/r/378/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241188
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Alec L Davis [Tue, 19 Jan 2010 00:24:35 +0000 (00:24 +0000)]
Update CDR variables before pbx starts (overlap dial)
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call.
(issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff2.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241187
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Jeff Peeler [Mon, 18 Jan 2010 22:31:25 +0000 (22:31 +0000)]
Extend max call limit duration from 24.8 days to 292+ million years.
If the limit was set past MAX_INT upon answering, the call was immediately
hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup).
The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been
changed to return an int64_t to prevent overflow. Also the reporter suggested
adding a message indicating the reason for the call hanging up. Given that the
new limit is so much higher, the message (which would only really be useful in
the overflow scenario) has been made a debug message only.
(closes issue #16006)
Reported by: viraptor
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241143
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Jason Parker [Mon, 18 Jan 2010 22:03:49 +0000 (22:03 +0000)]
Fix an RTP instance allocation failure on Solaris.
(closes issue #16543)
Reported by: crjw
Patches:
rtp_sin_family.patch uploaded by crjw (license 963)
Tested by: crjw, qwell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241098
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Alec L Davis [Mon, 18 Jan 2010 22:00:24 +0000 (22:00 +0000)]
Update CDR variables before pbx starts
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call.
(closes issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff.txt uploaded by alecdavis (license 585)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241097
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Sean Bright [Mon, 18 Jan 2010 19:57:52 +0000 (19:57 +0000)]
Merged revisions 241015 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r241015 | seanbright | 2010-01-18 14:54:19 -0500 (Mon, 18 Jan 2010) | 12 lines
Plug a memory leak when reading configs with their comments.
While reading through configuration files with the intent of returning their
full contents (comments specifically) we allocated some memory and then forgot
to free it. This doesn't fix 16554 but clears up a leak I had in the lab.
(issue #16554)
Reported by: mav3rick
Patches:
issue16554_20100118.patch uploaded by seanbright (license 71)
Tested by: seanbright
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241016
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Tilghman Lesher [Mon, 18 Jan 2010 19:26:07 +0000 (19:26 +0000)]
Make HASHes inheritable across channel creation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241012
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