https://origsvn.digium.com/svn/asterisk/branches/1.4
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r245792 | dvossel | 2010-02-09 16:55:38 -0600 (Tue, 09 Feb 2010) | 12 lines
Fixes iaxs and iaxsl size off by one issue.
2^15 = 32768 which is the maximum allowed iax2 callnumber.
Creating the iaxs and iaxsl array of size 32768 means the maximum
callnumber is actually out of bounds. This causes a nasty crash.
(closes issue #15997)
Reported by: exarv
Patches:
iax_fix.diff uploaded by dvossel (license 671)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After further discussion with Steve Underwood, we should not (yet) be offering
to receive MMR or JBIG transcoded streams from T.38 endpoints. A future spandsp
release will support those features, and then they can be enabled during
negotiation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
They were previously passed correctly, but they simply weren't used. This
caused issues with various platforms whose builds needed to pass special
linker flags via the configure script.
(closes issue #16596)
Reported by: pprindeville
Patches:
asterisk-1.6-astldflags.patch uploaded by pprindeville (license 347)
Tested by: tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This module needs to be converted to gtk2, or we will eventually have to just
remove it from the tree. gtk1 isn't even packaged anymore in the distro I'm
using. I suspect nobody uses this and that nobody would notice if we removed
it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
David Vossel
Tilghman Lesher
Matthew Nicholson
Kevin P. Fleming
Russell Bryant
Jason Parker
Olle Johansson
Mark Michelson
Jeff Peeler
Terry Wilson
Sean Bright
Richard Mudgett