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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245306 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245269 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245268 f38db490-d61c-443f-a65b-d21fe96a405b
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This config option is already handled by the function handle_common_options
and it is unnecessary to parse the value again.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245230 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@245192 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@245147 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@245090 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245046 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@245006 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16778)
Reported by: pitel
Patches:
diff.txt uploaded by twilson (license 396)
Tested by: twilson, pitel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244945 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244927 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244924 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244887 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244792 f38db490-d61c-443f-a65b-d21fe96a405b
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parse_moved_contact attempts to remove a quoted string
twice, and the first try wasn't even being done correctly.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244769 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244768 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244729 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244728 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244688 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244647 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244598 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244597 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244547 f38db490-d61c-443f-a65b-d21fe96a405b
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the first one.
(closes issue #16359)
Reported by: raarts
Patches:
dahdi-setvars.diff uploaded by raarts (license 937)
Tested by: raarts
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@244505 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244443 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244393 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244331 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244243 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244154 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@244071 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243986 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243943 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243860 f38db490-d61c-443f-a65b-d21fe96a405b
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243780 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243652 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243571 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16358)
Reported by: raarts
Patches:
lockconfdir.diff uploaded by raarts (license 937)
modified by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243551 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243487 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243482 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243391 f38db490-d61c-443f-a65b-d21fe96a405b
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Pushes code clean up done in app_externalivr back
into app_senddtmf
Review: https://reviewboard.asterisk.org/r/473/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243346 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243266 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243244 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243242 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243200 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@243118 f38db490-d61c-443f-a65b-d21fe96a405b
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functions are still broken.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243077 f38db490-d61c-443f-a65b-d21fe96a405b
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