This test works by reading input from arrays to build a sample
dialplan. From there, patterns are attempted to be matched against
said dialplan, with the expected match given. We then search in our
example dialplan to see if we find a match and if what we find matches
what we expected it to match.
(closes issue #16809)
Reported by: lmadsen
Tested by: mmichelson
Review: https://reviewboard.asterisk.org/r/504/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using ast_seekstream with the read/write streams of a monitor,
the number of samples we are seeking must be of the same rate as the
stream or the jump calculation will be incorrect. This patch adds logic
to correctly convert the number of samples to jump to the sample rate
the read/write stream is using.
For example, if the call is G722 (16khz) and the read/write stream is
recording a 8khz wav, seeking 320 samples of 16khz audio is not the
same as seeking 320 samples of 8khz audio when performing the ast_seekstream
on the stream.
ABE-2044
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change builds upon the recent change to menuselect to add 'touch_on_change'
as an attribute of both categories and members. This should allow only the most
invasive defines to cause a complete rebuild, while defines which only affect a
subset of modules will only cause a rebuild of that smaller set.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r246709 | tilghman | 2010-02-15 17:42:33 -0600 (Mon, 15 Feb 2010) | 5 lines
Make the menuselect instructions correct by allowing 'make menuselect' to actually solve dependency problems.
(Previously, it would fail out again with the same message about running
'make menuselect', which was NOT at all helpful.)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ooops. Failed to note that we were inside a for loop and
pri_channel_bridge() needs to be executed only once.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Code Refactoring Changes
- read_to_parts() moved to reqresp_parser.c and has been renamed as
get_name_and_number()
- get_in_brackets() moved to reqresp_parser.c
- find_closing_quotes() added to sip_utils.h
Logic Changes
- get_name_and_number() now uses parse_uri() and get_calleridname()
for parsing. Before this change only names within quotes were
found, when names not within quotes are possible.
New Unit Tests
-sip_get_name_and_number_test
-sip_get_in_brackets_test
(closes issue #16707)
Reported by: Nick_Lewis
Patches:
issue16706.diff uploaded by dvossel (license 671)
Review: https://reviewboard.asterisk.org/r/499/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r246545 | dvossel | 2010-02-12 17:30:17 -0600 (Fri, 12 Feb 2010) | 16 lines
lock channel during datastore removal
On channel destruction the channel's datastores are removed and
destroyed. Since there are public API calls to find and remove
datastores on a channel, a lock should be held whenever datastores are
removed and destroyed. This resolves a crash caused by a race
condition in app_chanspy.c.
(closes issue #16678)
Reported by: tim_ringenbach
Patches:
datastore_destroy_race.diff uploaded by tim ringenbach (license 540)
Tested by: dvossel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246546 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A patch was committed recently that converted duplicate uri parsing
code to use the parse_uri function. There were two instances where
this conversion did not mimic previous behavior exactly because the
port was not being parsed off the end of the domain. In order to do
this, a dummy pointer argument needs to be passed into parse_uri so
it will know it must parse out the port from the domain. If a port
output paramenter is not present, the domain is returned with the
port still attached.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This module includes a single test so far that creates events using two
different methods and does some verification on the result to make sure
the correct data can be retrieved from the event that was created.
One bug was found in the event API while developing this test, which makes
me happy. :-)
Review: https://reviewboard.asterisk.org/r/495/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/0009700@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 => 0009700,1,Wait(1) ;1 works, 3 did not
exten => 0009700,n,ChannelRedirect(${TESTCHAN},parkedcalls,701,1)
(closes issue #14992)
Reported by: davidw
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Detect all platforms that don't like that, either, and ensure that when documentation is
missing, we pass a non-NULL pointer when outputting the corresponding documentation.
(closes issue #16689)
Reported by: bklang
Patches:
20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/497/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1) It occurred to me that the difference in usage between the error ast_str and
the ast_test_update_status() usage has turned out to be a bit ambiguous in
practice. In a lot of cases, the same message was being sent to both.
In other cases, it was only sent to one or the other. My opinion now is that
in every case, I think it makes sense to do both; we should output it to the
CLI as well as save it off for logging purposes.
This change results in most of the changes in this diff, since it required
changes to all existing unit tests. It also allowed for some simplifications
of unit test API implementation code.
2) Update ast_test_status_update() to include the file, function, and line
number for the code providing the update.
3) There are some formatting tweaks here and there. Hopefully they aren't too
distracting for code review purposes. Reviewboard's diff viewer seems to do a
pretty good job of pointing out when something is a whitespace change.
4) I moved the md5_test and sha1_test into the test_utils module. It seemed
like a better approach since these tests are so tiny.
5) I changed the number of nodes used in heap_test_2 from 1 million to
100 thousand. The only reason for this was to reduce the time it took
for this test to run.
6) Remove an unused function prototype that was at the bottom of utils.h.
7) Simplify test_insert() using the LIST_INSERT_SORTALPHA() macro. The one
minor difference in behavior is that it no longer checks for a test registered
with the same name.
8) Expand the code in test_alloc() to provide specific error messages for each
failure case, to clearly inform developers if they forget to set the name,
summary, description, etc.
9) Tweak the output of the "test show registered" CLI command. I swapped the
name and category to have the category first. It seemed more natural since
that is the sort key.
10) Don't output the status ast_str in the "test show results" CLI command.
This is going to tend to be pretty verbose, so just leave that for the
detailed test logs (test generate results).
Review: https://reviewboard.asterisk.org/r/493/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@245864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
