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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237920 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236308 65c4cc65-6c06-0410-ace0-fbb531ad65f3