Mark Michelson [Mon, 22 Feb 2010 20:19:00 +0000 (20:19 +0000)]
Move the REF_DEBUG comment higher in the include list.
Uncommenting the REF_DEBUG definition where it was in the source
resulted in only a small part of the astobj2 references being logged
to a file. Moving this up higher in the include list causes all references
to be logged as they should be.
Make chan_misdn DTMF processing consistent with other channel technologies.
The processing of DTMF tones on the receiving side of an ISDN channel is
inconsistent with the way it is handled in other channels, especially
DAHDI analog. This causes DTMF tones sent from an ISDN phone to be
doubled at the connected party.
We are using the following 2 options of misdn.conf
1) astdtmf=yes
2) senddtmf=yes
Option one is necessary because the asterisk DSP DTMF detection is better
than mISDN's internal DSP. Not as many false positives.
Option two is necessary to transmit DTMF tones end to end when mISDN
channels are connected to SIP channels with out of band DTMF for example.
The symptom is that DTMF tones sent by an ISDN phone are doubled on the
way through asterisk when two mISDN channels are connected with a Local
channel in between or if it is bridged to an analog channel.
The doubling of DTMF tones is because DTMF is passed inband to asterisk by
the mISDN channel and passed out of band once again after the release of
the DTMF tone. Passing it inband is wrong. Neither an analog channel nor
SIP channel passes DTMF inband if configured to inband DTMF. Analog and
SIP channels filter out the DTMF tones because they use the voice frames
returned by ast_dsp_process. But chan_misdn passes the unfiltered input
voice frames instead.
To overcome one aspect of the problem, the doubling of DTMF tones when two
mISDN channels are directly bridged, someone made an 'optimization', where
in that case the DTMF tone passed out-of-band to the peer channel is not
translated to an inband tone at the transmit side. This optimization is
bad because it does not work in general. For example, analog channels or
mISDN channels when bridged through an intermediary local channel will
generate DTMF tones from out-of-band information. Also, of course, it
must not be done when there is no inband DTMF available.
This patch fixes the issue. Now chan_misdn will filter the received
inband DTMF signal the same as other channel types.
Another change included: No need to build an extra translation path
because ast_process_dsp does it if required.
Tilghman Lesher [Thu, 18 Feb 2010 21:42:53 +0000 (21:42 +0000)]
If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns.
NULL means the value is not specified for the column, which normally means
the driver uses whatever is the default value. However, on MySQL, placing
a NULL in either a float or integer column results in a retrieval of the 0
value. Hence, users get an errant error on load. This patch suppresses
that error and makes the value as if it was not there.
Note that this cannot be done in the realtime driver, because the lack of
difference between NULL and 0 can only be intepreted correctly by the
driver itself. If we did it in the realtime driver, then it would be
effectively impossible to set any realtime field to 0, because it would act
as if the field were unspecified and possibly take on a different value.
Richard Mudgett [Thu, 18 Feb 2010 18:31:44 +0000 (18:31 +0000)]
Fix placing ISDN calls on hold preventing native bridging from being reexamined after a transfer.
Consider the following scenario:
/-- B
A == * == Network
\-- C
Party B calls party A (EuroISDN BRI phone)
Party A puts B on hold using the HOLD/RETRIEVE messages.
Party A calls party C.
Party A puts C on hold to talk with party B again.
Party A transfers B to C by hanging up.
The call does not get the opportunity to get re-transferred into the ISDN
network by the native bridge because native bridging is not being
reexamined after the initial transfer.
Russell Bryant [Thu, 18 Feb 2010 04:20:11 +0000 (04:20 +0000)]
Merged revisions 247422 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r247422 | russell | 2010-02-17 22:19:01 -0600 (Wed, 17 Feb 2010) | 10 lines
Tweak argument handling for wget in the sounds Makefile.
1) Fix the check to see if we are using wget to not be full of fail. The
configure script populates this variable with the absolute path to wget if
it is found, so it didn't work.
2) Allow some extra arguments to be passed in for wget. This is just a simple
change to allow our Bamboo build script to tell wget to be quiet and not fill
up our logs with download status output.
........
Mark Michelson [Wed, 17 Feb 2010 21:22:40 +0000 (21:22 +0000)]
Fix two problems in ast_str functions found while writing a unit test.
1. The documentation for ast_str_set and ast_str_append state that
the max_len parameter may be -1 in order to limit the size of the
ast_str to its current allocated size. The problem was that the max_len
parameter in all cases was a size_t, which is unsigned. Thus a -1 was
interpreted as UINT_MAX instead of -1. Changing the max_len parameter
to be ssize_t fixed this issue.
2. Once issue 1 was fixed, there was an off-by-one error in the case
where we attempted to write a string larger than the current allotted
size to a string when -1 was passed as the max_len parameter. When trying
to write more than the allotted size, the ast_str's __AST_STR_USED was
set to 1 higher than it should have been. Thanks to Tilghman for quickly
spotting the offending line of code.
Oh, and the unit test that I referenced in the top line of this commit
will be added to reviewboard shortly. Sit tight...
Jeff Peeler [Wed, 17 Feb 2010 19:51:53 +0000 (19:51 +0000)]
Add support for GROUP_MATCH_COUNT regex matching on category
Current support for regex matching was previously only available on the group.
Also, error reporting for regex failures has been added. In addition to this
feature enhancement a unit test has been written to check the regular expression
logic to ensure the count operation is working as expected.
David Vossel [Wed, 17 Feb 2010 18:29:48 +0000 (18:29 +0000)]
addition of dynamic parkinglots feature
This feature allows for parkinglots to be created dynamically within
the dialplan. Thanks to all who were involved with getting this patch
written and tested!
Tilghman Lesher [Wed, 17 Feb 2010 06:25:15 +0000 (06:25 +0000)]
Make all of the various rtpqos parameters in this branch available from the CHANNEL function.
Also includes a test for retrieving rtpqos parameters, including a NULL RTP
driver. Additionally, some further separation of the SIP internal API into
headers was necessary.
Mark Michelson [Tue, 16 Feb 2010 23:44:33 +0000 (23:44 +0000)]
Add va_end calls to __ast_str_helper.
According to the man page for stdarg(3),
"Each invocation of va_copy() must be matched by a
corresponding invocation of va_end() in the same
function."
There were several cases in __ast_str_helper where
va_copy was not matched with a corresponding call
to va_end.
Mark Michelson [Tue, 16 Feb 2010 18:29:42 +0000 (18:29 +0000)]
Add unit test for dialplan pattern matching.
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.
David Vossel [Tue, 16 Feb 2010 17:07:41 +0000 (17:07 +0000)]
fixes sample rate conversion issue with Monitor application
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.
Tilghman Lesher [Tue, 16 Feb 2010 00:52:45 +0000 (00:52 +0000)]
Change the blanket rules to delete .lastclean on all CFLAGS menuselect targets to be more particular.
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.
David Vossel [Mon, 15 Feb 2010 15:45:02 +0000 (15:45 +0000)]
chan_sip parse code refactoring plus two new unit tests
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
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
........
David Vossel [Fri, 12 Feb 2010 17:59:39 +0000 (17:59 +0000)]
fixes areas where port should be removed from domain during parsing
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.
David Vossel [Thu, 11 Feb 2010 18:42:25 +0000 (18:42 +0000)]
astobj2 unit test and bug fix
A bug was discovered during the creation of the astobj2 unit test.
When OBJ_MULTIPLE | OBJ_UNLINK is used, the objects being returned
had a ref count issue. This patch resolves that.
Russell Bryant [Wed, 10 Feb 2010 23:19:16 +0000 (23:19 +0000)]
Add a test module for the event API, test_event.c.
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. :-)
Jeff Peeler [Wed, 10 Feb 2010 16:47:37 +0000 (16:47 +0000)]
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:
Tilghman Lesher [Wed, 10 Feb 2010 16:01:28 +0000 (16:01 +0000)]
Solaris doesn't like outputting a NULL to a %s in format strings.
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.
Russell Bryant [Tue, 9 Feb 2010 23:32:14 +0000 (23:32 +0000)]
Various updates to the unit test API.
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).
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.
Don't offer MMR or JBIG transcoding during T.38 negotiation.
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
Tilghman Lesher [Mon, 8 Feb 2010 22:31:40 +0000 (22:31 +0000)]
Actually use _ASTLDFLAGS in the main/ and channels/ Makefiles.
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.
Russell Bryant [Mon, 8 Feb 2010 04:43:55 +0000 (04:43 +0000)]
Add a todo for pbx_gtkconsole for updating to gtk2.
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.
Mark Michelson [Sat, 6 Feb 2010 14:43:03 +0000 (14:43 +0000)]
Remove useless sip options related to hash table size.
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.
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:
David Vossel [Thu, 4 Feb 2010 15:36:33 +0000 (15:36 +0000)]
fix truncated format string in 'test show registered'
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.
projects / thirdparty / asterisk.git / log
