SWP-1229
ABE-2161
* Ensure chan_local.c:local_call() will not leak cid.cid_dnid when
copying.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256104 f38db490-d61c-443f-a65b-d21fe96a405b
The ast_channel structure had a field used for linking a channel into a
linked list, but now that ast_channel structures are ao2 objects, this is
no longer needed, and could be harmful as ao2 objects really shouldn't
ever be placed into linked lists (since those lists don't assist with
reference count management on the objects).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254637 f38db490-d61c-443f-a65b-d21fe96a405b
application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254450 f38db490-d61c-443f-a65b-d21fe96a405b
users expect them to work.
'core set debug' and 'core set verbose' can optionally change the
level for a specific filename; however, this is actually for a
specific source file name, not the module that source file is included
in. With examples like chan_sip, chan_iax2, chan_misdn and others
consisting of multiple source files, this will not lead to the
behavior that users expect. If they want to set the debug level for
chan_sip, they want it set for all of chan_sip, and not to have to
also set it for reqresp_parser and other files that comprise the
chan_sip module.
This patch changes this functionality to be module-name based instead
of file-name based.
To make this work, some Makefile modifications were required to ensure
that the AST_MODULE definition is present in each object file produced
for each module as well.
Review: https://reviewboard.asterisk.org/r/574/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253917 f38db490-d61c-443f-a65b-d21fe96a405b
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@252089 f38db490-d61c-443f-a65b-d21fe96a405b
The mis-placement of the latest entry meant that when it was set, it was writing
one index past the end of the properties array in the ast_rtp_instance (which
happened to be the local_address field).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250871 f38db490-d61c-443f-a65b-d21fe96a405b
Only chan_dahdi set a value in cdrflags. Everyone else just copied it
around the system. Noone cared about any value it may have contained.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250565 f38db490-d61c-443f-a65b-d21fe96a405b
When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default. This value is required
in order for the adaptive jitterbuffer to work correctly. To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@249893 f38db490-d61c-443f-a65b-d21fe96a405b
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...
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247335 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@246260 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@246030 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@245864 f38db490-d61c-443f-a65b-d21fe96a405b
1. URI Encoding
This patch changes ast_uri_encode()'s behavior when doreserved is enabled.
Previously when doreserved was enabled only a small set of reserved
characters were encoded. This set was comprised primarily of the reserved
characters defined in RFC3261 section 25.1, but contained other characters as
well. Rather than only escaping the reserved set, doreserved now escapes
all characters not within the unreserved set as defined by RFC 3261 and
RFC 2396. Also, the 'doreserved' variable has been renamed to 'do_special_char'
in attempts to avoid confusion.
When doreserve is not enabled, the previous logic of only encoding the
characters <= 0X1F and > 0X7f remains, except for the '%' character, which
must always be encoded as it signifies a HEX escaped character during the decode
process.
2. URI Decoding: Break up URI before decode.
In chan_sip.c ast_uri_decode is called on the entire URI instead of it's
individual parts after it is parsed. This is not good as ast_uri_decode
can introduce special characters back into the URI which can mess up parsing.
This patch resolves this by not decoding a URI until parsing is completely
done. There are many instances where we check to see if pedantic checking
is enabled before we decode a URI. In these cases a new macro,
SIP_PEDANTIC_DECODE, is used on the individual parsed segments of the URI
rather than constantly putting if (pedantic) { decode() } checks everywhere
in the code. In the areas where ast_uri_decode is not dependent upon
pedantic checking this macro is not used, but decoding is still moved to
each individual part of the URI. The only behavior that should change from
this patch is the time at which decoding occurs.
Since I had to look over every place URI parsing occurs to create this
patch, I found several places where we use duplicate code for parsing.
To consolidate the code, those areas have updated to use the parse_uri()
function where possible.
3. SIP display-name decoding according to RFC3261 section 25.
To properly decode the display-name portion of a FROM header, chan_sip's
get_calleridname() function required a complete re-write. More information
about this change can be found in the comments at the beginning of this function.
4. Unit Tests.
Unit tests for ast_uri_encode, ast_uri_decode, and get_calleridname() have been
written. This involved the addition of the test_utils.c file for testing the
utils api.
(closes issue #16299)
Reported by: wdoekes
Patches:
astsvn-16299-get_calleridname.diff uploaded by wdoekes (license 717)
get_calleridname_rewrite.diff uploaded by dvossel (license 671)
Tested by: wdoekes, dvossel, Nick_Lewis
Review: https://reviewboard.asterisk.org/r/469/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243200 f38db490-d61c-443f-a65b-d21fe96a405b
This patch removes code that was duplicated from pbx.c to manager.c
in order to prevent API change in released versions of Asterisk.
There are propably also other places that would benefit from reading the
return code and react if a function returns error codes on writing a value into it.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242919 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242521 f38db490-d61c-443f-a65b-d21fe96a405b
If the limit was set past MAX_INT upon answering, the call was immediately
hung up due to overflow from the return of ast_tvdiff_ms (in ast_check_hangup).
The time calculation functions ast_tvdiff_sec and ast_tvdiff_ms have been
changed to return an int64_t to prevent overflow. Also the reporter suggested
adding a message indicating the reason for the call hanging up. Given that the
new limit is so much higher, the message (which would only really be useful in
the overflow scenario) has been made a debug message only.
(closes issue #16006)
Reported by: viraptor
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241143 f38db490-d61c-443f-a65b-d21fe96a405b
Specifically, by setting TESTTIME() to a particular date and time, you
can test whether a dialplan correctly branches as was intended. This was
developed after recent questions on the -users list on how to test their
holiday dialplan logic.
(closes issue #16464)
Reported by: tilghman
Patches:
20100112__issue16464.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/458/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239957 f38db490-d61c-443f-a65b-d21fe96a405b
During the process of removing an audiohook from one channel
and attaching it to another the audiohook's status is updated
to DONE and then back to whatever it was previously. Typically
updating the status after setting it to DONE is not a good idea
because DONE can trigger unrecoverable audiohook destruction
events... because of this a conditional check was added to
audiohook_update_status to explicitly prevent the audiohook
from ever changing after being set to DONE. It was this check
that prevented audiohook inherit from work properly though.
Now ast_audiohook_move_by_source is treated as a special exception,
as the audiohook must be returned to its previous status after
attaching it to the new channel. This is only a safe operation
because the audiohook's lock is held the entire time, otherwise
this could cause trouble.
(closes issue #16522)
Reported by: corruptor
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@238635 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@238091 f38db490-d61c-443f-a65b-d21fe96a405b
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: http://svn.digium.com/svn/asterisk/trunk@237406 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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: http://svn.digium.com/svn/asterisk/trunk@236613 f38db490-d61c-443f-a65b-d21fe96a405b
The Unit Test Framework is a new API that manages registration and
execution of unit tests in Asterisk with the purpose of verifying the
operation of C functions. The Framework consists of a single test
manager accompanied by a list of registered test functions defined
within the code. A test is defined, registered, and unregistered
from the framework using a set of macros which allow the test code
to only be compiled within asterisk when the TEST_FRAMEWORK flag is
enabled in menuselect. This allows the test code to exist in the
same file as the C functions it intends to verify. Registered tests
may be viewed and executed via a set of new CLI commands. CLI commands
are also present for generating and exporting test results into xml
and txt formats.
For more information and use cases please refer to the documentation
provided at the beginning of the test.h file.
Review: https://reviewboard.asterisk.org/r/447/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236027 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines
Correct CDR dispositions for BUSY/FAILED
This patch is simple in that it reorders the disposition defines so that the fix
for issue 12946 works properly (the default CDR disposition was changed to
AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to
ensure all CDR records are written.
The side effects of CDR changes are scary, so I'm documenting the test cases
performed to attempt to catch any regressions. The following tests were all
performed using 1.4 rev 195881 vs head (235571) + patch:
A calls B
C calls B (busy)
Hangup C
Hangup A
(Both SIP and features)
A calls B
A blind transfers to C
Hangup C
(Both SIP and features)
A calls B
A attended transfers to C
Hangup C
A calls B
A attended transfers to C (SIP)
C blind transfers to A (features)
Hangup A
All of the test scenario CDRs matched.
The following tests were performed just with the patch to ensure proper operation
(with unanswered=yes):
exten =>s,1,Answer
exten =>s,n,ResetCDR(w)
exten =>s,n,ResetCDR(w)
exten =>s,1,ResetCDR(w)
exten =>s,n,ResetCDR(w)
(closes issue #16180)
Reported by: aatef
Patches:
bug16180.patch uploaded by jpeeler (license 325)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235660 f38db490-d61c-443f-a65b-d21fe96a405b
The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@235342 f38db490-d61c-443f-a65b-d21fe96a405b
* Classes are now tracked past removal from the core container, and module
removal is actively prevented until all references are freed.
* A hanging reference stored in the channel has been removed. This could have
caused a mismatch and the music state not properly cleared, if two or more
reloads occurred between MOH being stopped and MOH being restarted.
* In certain circumstances, duplicate classes were possible.
* A race existed at reload time between a process being killed and the thread
responsible for reading from the related pipe respawning that process.
* Several reference counts have also been corrected. At least one could have
caused deleted classes to stick around forever, consuming resources. This
originally manifested as MOH external processes that were not killed at
reload time.
(closes issue #16279, closes issue #16207)
Reported by: parisioa, dcabot
Patches:
20091202__issue16279__2.diff.txt uploaded by tilghman (license 14)
Tested by: parisioa, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232660 f38db490-d61c-443f-a65b-d21fe96a405b
The manpage lies; it says that the function is in POSIX, but that's only for
ffs(3), not ffsll(3).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@232164 f38db490-d61c-443f-a65b-d21fe96a405b
In the process of swapping ULAW to a place in the extended codec space, we
found several unhandled cases, where a 32-bit integer was still being used to
handle a codec field. Most of these have been fixed with this commit, although
there is at least one case (codec_dahdi) which depends upon outside headers to
be altered before a conversion can be made.
(Fixes AST-278, SWP-459)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231850 f38db490-d61c-443f-a65b-d21fe96a405b
1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL
session, so that log/error/debug messages generated by the UDPTL stack can
be 'connected' to the endpoint that caused them to be generated.
2) Improve comments (and process) of calculating the far end's maximum IFP size
when redundancy mode is in use for error correction.
3) When an IFP larger than the calculated 'far max IFP' size is presented for
writing, truncate it rather than putting in the buffer and allowing the buffer
to overflow; this will cause the ends to retrain to a lower bit rate that
produces IFPs of an appropriate size if possible, and if not possible, the
FAX transfer will fail completely. In these cases, it is due to the one endpoint
supplying a T38FaxMaxDatagram value that is improperly calculated and is
too low to be of use; we have configuration options available to override
this behavior.
4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer
needed.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231692 f38db490-d61c-443f-a65b-d21fe96a405b
* Add REPLACE function, which searches a given variable for a set of
characters and replaces each with a given character.
* Add PASSTHRU function, which passes a literal string back, like a NoOp for
functions. Intent is to be able to specify a literal string to another
function that takes a variable name as an argument.
* Let the array manipulation functions work with dialplan functions, in
addition to variables. This allows the array manipulation functions to
modify ASTDB and ODBC backends, assuming the func_odbc configuration has
both read and write functions.
(closes issue #15223)
Reported by: ajohnson
Patches:
20091112__issue15223.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@230994 f38db490-d61c-443f-a65b-d21fe96a405b
This patch, originally submitted by jozza, enables custom modules to send actions to AMI
and receive messages from AMI via a hook interface. Included is a simple test module to
illustrate the interface.
(closes issue #14635)
Reported by: jozza
Review: https://reviewboard.asterisk.org/r/412/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227448 f38db490-d61c-443f-a65b-d21fe96a405b
This is a side project I've been poking at this week. The intent is to discuss
Asterisk architecture in a top down fashion to help new developers understand how
Asterisk is put together. There is a ton of stuff to write about, so this will
just continue to evolve over time.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226606 f38db490-d61c-443f-a65b-d21fe96a405b
* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension])
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
* Added support for BRI PTMP NT mode.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225692 f38db490-d61c-443f-a65b-d21fe96a405b
Added documentation on how to create a local git repository from
SVN. This documentation was added via doxygen.
(closes issue #15814)
Reported by: tzafrir
Patches:
git-asterisk-howto uploaded by tzafrir (license 46)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225483 f38db490-d61c-443f-a65b-d21fe96a405b
What this patch fixes
1.Moves sip TCP/TLS connection setup into the TCP helper thread:
Connection setup takes awhile and before this it was being
done while holding the monitor lock.
2.Moves TCP/TLS writing to the TCP helper thread: Through the
use of a packet queue and an alert pipe, the TCP helper thread
can now be woken up to write data as well as read data.
3.Locking error: sip_xmit returned an XMIT_ERROR without giving
up the tcptls_session lock. This lock has been completely removed
from sip_xmit and placed in the new sip_tcptls_write() function.
4.Memory leak: When creating a tcptls_client the tls_cfg was alloced
but never freed unless the tcptls_session failed to start. Now the
session_args for a sip client are an ao2 object which frees the
tls_cfg on destruction.
5.Pointer to stack variable: During sip_prepare_socket the creation
of a client's ast_tcptls_session_args was done on the stack and
stored as a pointer in the newly created tcptls_session. Depending
on the events that followed, there was a slight possibility that
pointer could have been accessed after the stack returned. Given
the new changes, it is always accessed after the stack returns
which is why I found it.
Notable code changes
1.I broke tcptls.c's ast_tcptls_client_start() function into two
functions. One for creating and allocating the new tcptls_session,
and a separate one for starting and handling the new connection.
This allowed me to create the tcptls_session, launch the helper
thread, and then establish the connection within the helper thread.
2.Writes to a tcptls_session are now done within the helper thread.
This is done by using an alert pipe to wake up the thread if new
data needs to be sent. The thread's sip_threadinfo object contains
the alert pipe as well as the packet queue.
3.Since the threadinfo object contains the alert pipe, it must now be
accessed outside of the helper thread for every write (queuing of a
packet). For easy lookup, I moved the threadinfo objects from a
linked list to an ao2_container.
(closes issue #13136)
Reported by: pabelanger
Tested by: dvossel, whys
(closes issue #15894)
Reported by: dvossel
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/380/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225445 f38db490-d61c-443f-a65b-d21fe96a405b
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing
"desk to desk" between offices each with an asterisk box over the ISDN
should then be possible, without a whole load of DDI numbers required.
(closes issue #15604)
Reported by: alecdavis
Patches:
asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585)
Some minor modificatons were made.
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/405/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225357 f38db490-d61c-443f-a65b-d21fe96a405b
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.
During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.
Review: https://reviewboard.asterisk.org/r/379/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225244 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines
Isolate frames returned from a DSP instance or codec translator.
The reasoning for these changes are the same as what I wrote in the commit
message for rev 222878.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224932 f38db490-d61c-443f-a65b-d21fe96a405b
Two examples of its use are included, and the usage could be expanded in some
cases into certain configuration options where time periods are specified.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224225 f38db490-d61c-443f-a65b-d21fe96a405b
This isn't the best way to do this, but it is the easiest. There are some
limitations that are going to need to be addressed at some point with reloads
and when I (or someone else) work on that, then the API can be updated to
handle passing the private config data that the calendar tech modules need in
a better way as well.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@223016 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines
Make filestream frame handling safer by isolating frames before returning them.
This patch is related to a number of issues on the bug tracker that show
crashes related to freeing frames that came from a filestream. A number of
fixes have been made over time while trying to figure out these problems, but
there re still people seeing the crash. (Note that some of these bug reports
include information about other problems. I am specifically addressing
the filestream frame crash here.)
I'm still not clear on what the exact problem is. However, what is _very_
clear is that we have seen quite a few problems over time related to unexpected
behavior when we try to use embedded frames as an optimization. In some cases,
this optimization doesn't really provide much due to improvements made in other
areas.
In this case, the patch modifies filestream handling such that the embedded frame
will not be returned. ast_frisolate() is used to ensure that we end up with a
completely mallocd frame. In reality, though, we will not actually have to malloc
every time. For filestreams, the frame will almost always be allocated and freed
in the same thread. That means that the thread local frame cache will be used.
So, going this route doesn't hurt.
With this patch in place, some people have reported success in not seeing the
crash anymore.
(SWP-150)
(AST-208)
(ABE-1834)
(issue #15609)
Reported by: aragon
Patches:
filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2)
Tested by: aragon, russell
(closes issue #15817)
Reported by: zerohalo
Tested by: zerohalo
(closes issue #15845)
Reported by: marhbere
Review: https://reviewboard.asterisk.org/r/386/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222880 f38db490-d61c-443f-a65b-d21fe96a405b
Channels are stored in an ao2_container. When accessing an item within
an ao2_container the proper locking order is to first lock the container,
and then the items within it.
In ast_do_masquerade both the clone and original channel must be locked
for the entire duration of the function. The problem with this is that
it attemptes to unlink and link these channels back into the ao2_container
when one of the channel's name changes. This is invalid locking order as
the process of unlinking and linking will lock the ao2_container while
the channels are locked!!! Now, both the channels in do_masquerade are
unlinked from the ao2_container and then locked for the entire function.
At the end of the function both channels are unlocked and linked back
into the container with their new names as hash values.
This new method of requiring all channels and tech pvts to be unlocked
before ast_do_masquerade() or ast_change_name() required several
changes throughout the code base.
(closes issue #15911)
Reported by: russell
Patches:
masq_deadlock_trunk.diff uploaded by dvossel (license 671)
Tested by: dvossel, atis
(closes issue #15618)
Reported by: lmsteffan
Patches:
deadlock_local_attended_transfers_trunk.diff uploaded by dvossel (license 671)
Tested by: lmsteffan, dvossel
Review: https://reviewboard.asterisk.org/r/387/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222761 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines
Fix ao2_iterator API to hold references to containers being iterated.
See Mantis issue for details of what prompted this change.
Additional notes:
This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK
has become an enum instead of a macro, with a name that fits our
naming policy; also, it is now necessary to call
ao2_iterator_destroy() on any iterator that has been
created. Currently this only releases the reference to the container
being iterated, but in the future this could also release other
resources used by the iterator, if the iterator implementation changes
to use additional resources.
(closes issue #15987)
Reported by: kpfleming
Review: https://reviewboard.asterisk.org/r/383/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@222176 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221266 f38db490-d61c-443f-a65b-d21fe96a405b
JABBER_RECEIVE (along with JabberSend) makes Asterisk interact with users over
XMPP to process calls.
SendText can be used instead of JabberSend in the context of XMPP based voice
channels (chan_gtalk and chan_jingle).
(closes issue #12569)
Reported by: eech55
Tested by: phsultan, asannucci, lmadsen, jtodd, maxgo
Review: https://reviewboard.asterisk.org/r/88/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220457 f38db490-d61c-443f-a65b-d21fe96a405b
Most of the functionality here is gained simply by setting the feature flag
on the bridge config. However, the dial limit functionality has been moved from
app_dial to the features code and has been made public so both app_dial and
the bridge app can use it.
(closes issue #13165)
Reported by: tim_ringenbach
Patches:
app_bridge_options_r138998.diff uploaded by tim ringenbach (license 540),
modified by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@220344 f38db490-d61c-443f-a65b-d21fe96a405b
This commit adds the doxygen changes that I've made to describe the Mantis
work flow documentation for the open source issue tracker. This should make
it easier to determine the flow of issues through the issue tracker, and what
those statuses mean.
(closes issue #15902)
Reported by: lmadsen
Patches:
mantisworkflow.h uploaded by lmadsen (license 10)
Review: https://reviewboard.asterisk.org/r/367/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219895 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines
Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down.
This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h.
(closes issue #15316)
Reported by: vmarrone
Tested by: mnicholson
Review: https://reviewboard.asterisk.org/r/362/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@219139 f38db490-d61c-443f-a65b-d21fe96a405b
Most importantly, note that a NULL description will cause a
crash, as I just experienced that firsthand.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217158 f38db490-d61c-443f-a65b-d21fe96a405b
A recent change to the configure script that allows the user to specify
CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of
letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build
system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That
problem is now corrected.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@217074 f38db490-d61c-443f-a65b-d21fe96a405b
Added detection of DTMF tone energy levels on FXO channels in chan_dahdi
monitoring loop so DTMF CID can be detected without the need of a polarity
change precursor.
(closes issue #9096)
Reported by: fleed
Patches:
9096-chan_dahdi-trunk.diff uploaded by dbailey (license 819)
Tested by: cyberplant, sum, maturs
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@216094 f38db490-d61c-443f-a65b-d21fe96a405b
This did not function in the way that was intended, causing more compatibility
issues than it solved. It is best, therefore, that it be simply removed.
(Discussed with kpfleming; agreement to remove was reached.)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215800 f38db490-d61c-443f-a65b-d21fe96a405b
One note on defining _POSIX_C_SOURCE: while this feature test macro
works to require certain behaviors on Linux, it works differently on *BSD
platforms to REMOVE certain API calls that are not in the POSIX specification,
such as vasprintf(3). Thus, defining it while depending upon vasprintf (and
other extensions to the POSIX standard) to be defined is a recipe to ensure
that Asterisk is only buildable on Linux.
Hence, this define which was meant to INCREASE portability, effectively
ensures the opposite.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@214863 f38db490-d61c-443f-a65b-d21fe96a405b
Cross-compilation environments want to provide 'defaults' for compiler and
linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the
environment or as command-line arguments to the configure script. This patch
modifies the configure script and Makefile to preserve these settings and
ensure they are used in the build process.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@214696 f38db490-d61c-443f-a65b-d21fe96a405b
It's not clear that we actually require XOPEN_VERSION to be 600 or greater
at this time, so skip the check for now.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212672 f38db490-d61c-443f-a65b-d21fe96a405b
Based on a post on the gcc-help mailing list and some subsequent reading,
we can increase our portability to various platforms by directly defining
the POSIX and X/OPEN API feature sets we wish to have available. This patch
does that, and also includes a double-check to ensure that the system
we are compiling on can actually provide the requested feature sets.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212463 f38db490-d61c-443f-a65b-d21fe96a405b
This patch makes some small changes to handle watchdog timeouts in a better way,
and also uses a 'cleaner' method of including the spandsp header files.
(closes issue #14769)
Reported by: andrew
Patches:
app_fax-20090406.diff uploaded by andrew (license 240)
v1-14769.patch uploaded by dimas (license 88)
Tested by: freh, deti, caspy, dimas, sgimeno, Dovid
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210777 f38db490-d61c-443f-a65b-d21fe96a405b
The udptl-updates branch that was merged yesterday failed to properly send back
T.38 SDP responses with the correct error correction mode, if the incoming SDP
from the other end caused us to change error correction modes. This patch
corrects that situation.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208548 f38db490-d61c-443f-a65b-d21fe96a405b
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208464 f38db490-d61c-443f-a65b-d21fe96a405b
This commit introduces the security events API. This API is to be used by
Asterisk components to report events that have security implications.
A simple example is when a connection is made but fails authentication. These
events can be used by external tools manipulate firewall rules or something
similar after detecting unusual activity based on security events.
Inside of Asterisk, the events go through the ast_event API. This means that
they have a binary encoding, and it is easy to write code to subscribe to these
events and do something with them.
One module is provided that is a subscriber to these events - res_security_log.
This module turns security events into a parseable text format and sends them
to the "security" logger level. Using logger.conf, these log entries may be
sent to a file, or to syslog.
One service, AMI, has been fully updated for reporting security events.
AMI was chosen as it was a fairly straight forward service to convert.
The next target will be chan_sip. That will be more complicated and will
be done as its own project as the next phase of security events work.
For more information on the security events framework, see the documentation
generated from doc/tex/. "make asterisk.pdf"
Review: https://reviewboard.asterisk.org/r/273/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206021 f38db490-d61c-443f-a65b-d21fe96a405b
Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).
This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.
(closes issue #14849)
Reported by: afosorio
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205696 f38db490-d61c-443f-a65b-d21fe96a405b
While doing some reading about OpenSSL, I noticed a couple of things that
needed to be improved with our usage of OpenSSL.
1) We had initialization of the library done in multiple modules. This has now
been moved to a core function that gets executed during Asterisk startup.
We already link OpenSSL into the core for TCP/TLS functionality, so this
was the most logical place to do it.
2) OpenSSL is not thread-safe by default. However, making it thread safe is
very easy. We just have to provide a couple of callbacks. One callback
returns a thread ID. The other handles locking. For more information,
start with the "Is OpenSSL thread-safe?" question on the FAQ page of
openssl.org.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@205120 f38db490-d61c-443f-a65b-d21fe96a405b
Also go back and wrap all of the places that use the specific reverse charge
APIs with preprocessor conditionals.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204919 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines
Improved mapping of extension states from combined device states.
This fixes a few issues with incorrect extension states and adds
a cli command, core show device2extenstate, to display all possible
state mappings.
(closes issue #15413)
Reported by: legart
Patches:
exten_helper.diff uploaded by dvossel (license 671)
Tested by: dvossel, legart, amilcar
Review: https://reviewboard.asterisk.org/r/301/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204710 f38db490-d61c-443f-a65b-d21fe96a405b
Someone asked yesterday, "is there a good reason why we can't just put these
modules in Asterisk?". After a brief discussion, as long as the modules are
clearly set aside in their own directory and not enabled by default, it is
perfectly fine.
For more information about why a module goes in addons, see README-addons.txt.
chan_ooh323 does not currently compile as it is behind some trunk API updates.
However, it will not build by default, so it should be okay for now.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204413 f38db490-d61c-443f-a65b-d21fe96a405b
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
Also change the default casing of the string contants to lowercase. This really
just saves us from have to lowercase them later when displaying them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203605 f38db490-d61c-443f-a65b-d21fe96a405b
Modules placed in the priority heap for loading were not properly removed from the linked list. This resulted in some modules attempting to load twice.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@202410 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines
Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.
When the list to be appended is empty, and the list to be appended to is *not*,
AST_LIST_APPEND_LIST would actually cause the target list to become broken,
and no longer have a pointer to its last entry. This patch fixes the problem.
(reported by Stanislaw Pitucha on the asterisk-dev mailing list)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201262 f38db490-d61c-443f-a65b-d21fe96a405b
Some applications (notably app_fax) do not need digit detection nor FAX tone
detection while they are running, and if Asterisk is using software DSPs to provide
the detection, this consumes extra CPU cycles that could be better spent on the
actual application. This patch allows applications to query and control the state
of digit and tone detection on a channel, and modifies app_fax to disable them
while the FAX operations are occurring (and re-enable digit detection afterwards).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201139 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@201056 f38db490-d61c-443f-a65b-d21fe96a405b
Fix up modules in the 'apps' directory, and also correct the bad example of
enum definitions in include/asterisk/app.h, which many developers followed
(thanks for reading the documentation!). In addition, add some basic usage
examples of the 'pahole' and 'pglobal' tools to the coding guidelines.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200656 f38db490-d61c-443f-a65b-d21fe96a405b
This patch provides a new implementation of the optional API support defined
in asterisk/optional_api.h; this new version provides solves compatibility
issues with the use of linker version scripts for suppressing global symbols.
In addition, there is now a functional (and tested!) implementation for Mac OS/X,
so module writers no longer need to use special tests before calling optional
API functions. All future implementations must provide these same semantics,
so that module writers can rely on them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200519 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority
on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty.
(closes issue #15191)
Reported by: alecdavis
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/262/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199743 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines
Increase the size of our thread stack on 64 bit processors.
We were setting the stack size for each thread to 240KB regardless of
architecture, which meant that in some scenarios we actually had less available
stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we
calculate the stack size we reserve based on the platform's __WORDSIZE, which
gives us:
32 bit -> 240KB
64 bit -> 496KB
128 bit -> 1008KB (that's right, we're ready for 128 bit processors)
Patch typed by me but written by several members of #asterisk-dev, including
Kevin, Tilghman, and Qwell.
(closes issue #14932)
Reported by: jpiszcz
Patches:
06052009_issue14932.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines
Fix a typo in the stack size calculation just introduced.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199630 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines
Fixes issue with hints giving unexpected results.
Hints with two or more devices that include ONHOLD gave unexpected results.
(closes issue #15057)
Reported by: p_lindheimer
Patches:
onhold_trunk.diff uploaded by dvossel (license 671)
pbx.c.1.4.patch uploaded by p (license 558)
devicestate.c.trunk.patch uploaded by p (license 671)
Tested by: p_lindheimer, dvossel
Review: https://reviewboard.asterisk.org/r/254/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199298 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
Safely handle AMI connections/reload requests that occur during startup.
During asterisk startup, a lock on the list of modules is obtained by the
primary thread while each module is initialized. Issue 13778 pointed out a
problem with this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a connected client
(via Action: Command), causing a deadlock.
The resolution for 13778 was to move initialization of the manager to happen
after the other modules had already been lodaded. While this fixed this
particular issue, it caused a problem for users (like FreePBX) who call AMI
scripts via an #exec in a configuration file (See issue 15189).
The solution I have come up with is to defer any reload requests that come in
until after the server is fully booted. When a call comes in to
ast_module_reload (from wherever) before we are fully booted, the request is
added to a queue of pending requests. Once we are done booting up, we then
execute these deferred requests in turn.
Note that I have tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded, and if a
general reload request comes in ('module reload') the queue is flushed and we
only issue a single deferred reload for the entire system.
As for how this will impact existing installations - Before 13778, a reload
issued before module initialization was completed would result in a deadlock.
After 13778, you simply couldn't connect to the manager during startup (which
causes problems with #exec-that-calls-AMI configuration files). I believe this
is a good general purpose solution that won't negatively impact existing
installations.
(closes issue #15189)
(closes issue #13778)
Reported by: p_lindheimer
Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer, seanbright
Review: https://reviewboard.asterisk.org/r/272/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199051 f38db490-d61c-443f-a65b-d21fe96a405b
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198856 f38db490-d61c-443f-a65b-d21fe96a405b
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198727 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines
Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition.
This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels.
(closes issue #12946)
Reported by: meral
Patches:
null-cdr2.diff uploaded by mnicholson (license 96)
Tested by: mnicholson, dbrooks
(closes issue #15122)
Reported by: sum
Tested by: sum
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198072 f38db490-d61c-443f-a65b-d21fe96a405b
This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS
Exchange calendars. Exchange support has only been tested on Exchange Server 2k3
and does not support forms-based authentication at this time (patches *very*
welcome). Exchange support is also currently missing the ability to return a
list of a meting's attendees (again, patches are very, very welcome).
Features include:
Querying a calendar for events over a specific time range
Checking a calendar's busy status via the dialplan
Writing calendar events via the dialplan (CalDAV and Exchange only)
Handling calendar event notifications through the dialplan
(closes issue #14771)
Tested by: lmadsen, twilson, Shivaprakash
Review: https://reviewboard.asterisk.org/r/58
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197738 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197543 f38db490-d61c-443f-a65b-d21fe96a405b
'channel originate ... application <app>' CLI command.
(And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop,
wanna fight about it!?)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196758 f38db490-d61c-443f-a65b-d21fe96a405b
A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196308 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines
This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases.
This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected.
This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times.
(closes issue #13797)
Reported by: sh0t
Tested by: sh0t
(closes issue #14744)
Reported by: deepesh
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@195882 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds the ability for modules to dynamically create logger levels for their own use; these are named levels just like the built-in levels, and can be directed to any destination that the logger can send any level to, by including their names in logger.conf.
Review: https://reviewboard.asterisk.org/r/244/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194610 f38db490-d61c-443f-a65b-d21fe96a405b
This commit adds an API call that can be used to allocate a structure along with this stringfield storage in a single allocation.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192362 f38db490-d61c-443f-a65b-d21fe96a405b
Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192357 f38db490-d61c-443f-a65b-d21fe96a405b
As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192318 f38db490-d61c-443f-a65b-d21fe96a405b
This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192279 f38db490-d61c-443f-a65b-d21fe96a405b
This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@192059 f38db490-d61c-443f-a65b-d21fe96a405b
This patch add a new payload type for information elements, a set
of bit flags. The payload is transported as a 32-bit unsigned integer
but when matching is performed between events and subscribers,
the matching is done by using a bitwise AND instead of numeric value
comparison.
Review: http://reviewboard.asterisk.org/r/242/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191919 f38db490-d61c-443f-a65b-d21fe96a405b
In discussions today at the Europe Asterisk Developer Meet-Up, we determined that
the event_log was used in only 9 places in the entire tree, and really was not needed
at all. The users have been converted to use LOG_NOTICE, or the messages have been
removed since other messages were already in place that provided the same information.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191785 f38db490-d61c-443f-a65b-d21fe96a405b
Adds ability for CHANNEL() dialplan function, when used on DAHDI channels,
to temporarily change the number of buffers and/or the buffer policy, and also
to enable, disable, or switch the echo canceller between FAX/data and voice
modes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191411 f38db490-d61c-443f-a65b-d21fe96a405b
chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip.
(closes issue #14770)
Reported by: TheOldSaint
(closes issue #14768)
Reported by: TheOldSaint
Review: http://reviewboard.digium.com/r/240/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191177 f38db490-d61c-443f-a65b-d21fe96a405b
This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result. No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191140 f38db490-d61c-443f-a65b-d21fe96a405b
ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though.
Review: http://reviewboard.digium.com/r/237/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191028 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r190721 | kpfleming | 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) | 7 lines
Fix 'inconsistent line endings' when autoconf 2.63 is used
Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings
This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190725 f38db490-d61c-443f-a65b-d21fe96a405b