https://origsvn.digium.com/svn/asterisk/branches/1.10
................
r328611 | markm | 2011-07-18 08:56:49 -0400 (Mon, 18 Jul 2011) | 15 lines
Merged revisions 328608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | 9 lines
If the sip private structure is null, sip_setoption() will defref the null pointer and crash.
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328612 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.10
........
r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 Jul 2011) | 15 lines
Preserve sample rate quality of wideband mixmonitor recordings.
MixMonitor has the ability to record in any file format Asterisk supports,
but the quality of wideband audio is not preserved. This is because
regardless of the sample rate the call is being recorded in, the audio
is always downsampled to 8khz and then upsampled to whatever wideband
format it is being written as. This patch resolves this by requesting
the audio from the audiohook in the signed linear format closest to the
sample rate of the format we are writing. This fix is only possible for
Asterisk 1.10 because audio hooks in 1.8 are not capable of wideband
audio.
Review: https://reviewboard.asterisk.org/r/1314/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328121 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul 2011) | 14 lines
Correct double-free situation in manager output processing.
The process_output() function calls ast_str_append() and xml_translate() on its
'out' parameter, which is a pointer to an ast_str buffer. If either of these
functions need to reallocate the ast_str so it will have more space, they will
free the existing buffer and allocate a new one, returning the address of the
new one. However, because process_output only receives a pointer to the ast_str,
not a pointer to its caller's variable holding the pointer, if the original
ast_str is freed, the caller will not know, and will continue to use it (and
later attempt to free it).
(reported by jkroon on #asterisk-dev)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327953 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 Jul 2011) | 12 lines
Added additional checks for mailbox / password beginning with '*' character
A bug existed such that if a user entered a password with '*', and the extension 'a' did not exist, an invalid mailbox would be created and the user authenticated. The code was changed to prevent this from occurring, and to prevent users from having mailboxes or passwords defined that begin with the '*' character.
(closes issue ASTERISK-17443)
Reported by: Kevin Scott Adams
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1316/
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327856 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) | 14 lines
Use 'printf' (POSIX issue 4) instead of 'echo -n', for portability.
The problem with using 'echo -n' is that it is not portable. While BSD systems
required that the '-n' option be removed and interpreted, System V required
that all strings should be echoed with no interpretation of options. This
fundamental difference of behavior means that it is never possible to use the
'-n' flag to echo in tests which are meant to be portable.
In this case, on Mac OS X 10.6, the /bin/sh shell builtin 'echo' uses the
System V semantics of the command, and thus the SHELL test failed on that
platform.
http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327794 f38db490-d61c-443f-a65b-d21fe96a405b
When undergoing a shutdown and channels are kicked out of a bridge, a segfault
occurs because ConfBridge tries to play sounds on the bridge after the
underlying channels have been blown away due to the shutdown.
(closes ASTERISK-18040)
Review: https://reviewboard.asterisk.org/r/1283/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327748 f38db490-d61c-443f-a65b-d21fe96a405b
follow_talker mode originally echoed the same video stream
to all participants. As the primary talker switched around, the
video stream would result in the talker seeing themselves. Now
the primary talker sees the last person who was talking rather than
themselves.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327640 f38db490-d61c-443f-a65b-d21fe96a405b
generating party time to send its own T.38 reinvite.
Also don't forward frames through the gateway if we are negotiating T.38.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327511 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) | 9 lines
INVITE 403 Forbidden response always retransmits the maximum times.
Asterisk sends a 403 Forbidden response if authentication fails for an
INVITE as required. However, it ignores the ACK and keeps retransmitting
the response.
* Made not delete the to-tag in the dialog so the expected ACK can be
matched with the dialog and stop the retransmissions.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327212 f38db490-d61c-443f-a65b-d21fe96a405b
It was inconsistent to have the silk and celt format attribute
modules in the format file interpreter folder.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327116 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul 2011) | 11 lines
Reset our ast_str before passing it on to dialplan function backends.
It is possible for a dialplan backend to not modify the given buffer or ast_str
and still return success. This causes any previous value stored in the buffer
to be used as if the new function call provided it. Some functions also append
to the given buffer assuming it is empty.
The test_substitution unit test has also been modified to detect this problem.
(closes issue ASTERISK-17878)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327107 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) | 12 lines
Some code cleanup in pbx.c
* Mostly comment and format changes.
* ast_context_remove_extension_callerid() and ast_add_extension_nolock()
will write lock the found specific context.
* ast_context_find() will now tolerate a NULL name.
* Eliminated some inlined versions of find_context() and
find_context_locked().
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@327000 f38db490-d61c-443f-a65b-d21fe96a405b
markm