From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From Review Board:
There are two interrelated changes here.
First, there is the introduction of func_srv. This adds two new read-only
dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the
ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV
records instead. In order to facilitate this work, I added a couple of new API
calls to srv.h. ast_srv_get_record_count tells the number of records returned
by an SRV lookup. This number is calculated at the time of the SRV lookup.
ast_srv_get_nth_record allows one to get a numbered SRV record.
Second, there is the modification to chan_sip that allows one to specify a
hostname or IP address (along with a port) to send an outgoing INVITE to when
dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV
records and then use the host and port from the results to dial via a specific
host instead of what is configured in sip.conf.
Review: https://reviewboard.asterisk.org/r/608
SWP-1200
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Some platforms prefix externally-visible symbols in object files generated
from C sources (most commonly, '_' is the prefix). On these platforms,
the existing symbol export filtering process ends up suppressing all the symbols
that are supposed to be left visible. This patch allows the prefix string
to be supplied to the top-level Makefile in the LINKER_SYMBOL_PREFIX variable,
and then generates the linker scripts as required to include the prefix
supplied.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@255906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Here is a cut and paste of my review request for this change:
This past weekend, Russell ran our current suite of unit tests for Asterisk
under valgrind. The PBX pattern match test caused valgrind to spew forth two
invalid read errors. This patch contains two changes that shut valgrind up and
do not cause any new memory leaks.
Change 1: In ast_context_remove_extension_callerid2, valgrind reported an
invalid read in the for loop close to the function's end. Specifically, one of
the the strcmp calls in the loop control was reading invalid memory. This was
because the caller of ast_context_remove_extension_callerid2 (__ast_context
destroy in this case) passed as a parameter a shallow copy of an ast_exten's
exten field. This same ast_exten was what was destroyed inside the for loop,
thus any iterations of the for loop beyond the destruction of the ast_exten
would result in invalid reads. My fix for this is to make a copy of the
ast_exten's exten field and pass the copy to
ast_context_remove_extension_callerid2. In addition, I have also acted
similarly with the ast_exten's matchcid field. Since in this case a NULL is
handled quite differently than an empty string, I needed to be a bit more
careful with its handling.
Change 2: In __ast_context_destroy, we iterated over a hashtab and called
ast_context_remove_extension_callerid2 on each item. Specifically, the hashtab
over which we were iterating was an ast_exten's peer_table. Inside of
ast_context_remove_extension_callerid2, we could possibly destroy this
ast_exten, which also caused the hashtab to be freed. Attempting to call
ast_hashtab_end_traversal on the hashtab iterator caused an invalid read to
occur when trying to read the iterator->tab->do_locking field since
iterator->tab had already been freed. My handling of this problem is a bit less
straightforward. With each iteration over the hashtab's contents, we set a
variable called "end_traversal" based on the return of
ast_context_remove_extension_callerid2. If 0 is ever returned, then we know
that the extension was found and destroyed. Because of this, we cannot call
ast_hashtab_end_traversal because we will be guaranteeing a read of invalid
memory. In such a case, we forego calling ast_hashtab_end_traversal and instead
call ast_free on the hashtab iterator.
Review: https://reviewboard.asterisk.org/r/585
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This changes native bridging to break one millisecond early so that the more
accurate timeval calculations done in the generic bridge can be performed using
the bridge config. Currently the time between exiting native bridging slightly
late can sometimes cause a large enough discrepancy for warnings to be missed.
For the record, 1.4 does not attempt to native bridge at all when warnings are
enabled.
(closes issue #15815)
Reported by: adomjan
Review: https://reviewboard.asterisk.org/r/577/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We can have bad results when a module, upon being loaded, attempts
to reference the channels container if the container hasn't yet
been initialized. I saw this happen by trying to preload pbx_config.so
and having a hint defined which referenced a non-existent SIP peer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
internal_ao2_ref uses INTERNAL_OBJ which mzy report 'bad magic number', but
internal_ao2_ref continues on, causing segfault.
Although AO2_MAGIC number is checked by INTERNAL_OBJ before internal_ao2_ref is
called, A02_MAGIC is being destroyed (or a wrong pointer) by the time
internal_ao2_ref uses INTERNAL_OBJ.
internal_ao2_ref now returns -1 if INTERNAL_OBJ encouters a bad magic number.
(issue #17037)
Reported by: alecdavis
Patches:
bug17037.diff.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On Linux (glibc), regcomp() does not return an error for an empty string.
However, the version on OSX will return an error. The test for channel group
matching by regex now passes on the mac, as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252241 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(Copied from reviewboard)
Tests the following:
1. Basic allocation and setting of string fields.
2. Shrinking a string field and re-expanding it.
3. Growing the last allocation in a string field pool.
4. Setting a string to a large value such that a new string field pool must be
allocated.
In each part, we make sure that the string field is accurate (has the correct
value in it), make sure that the 2 bytes before the string field has the correct
capacity for the field, and for tests 2-4, we make sure that the string field is
where we expect it to be in memory.
Also tested:
5. Shrinking a string field and partially re-expanding it.
6. Setting strings in such a way as to create three separate string field pools
and then removing the middle pool.
There is a bug fix in the init function, which ensures the embedded_pool is set
to NULL which is important for stack allocated structures.
Review: https://reviewboard.asterisk.org/r/185/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The patch ensures that if a peer does not exist, parking settings are read from
the channel. A unit test has been written to ensure proper operation for both
standard parking and parking using masquerades.
(closes issue #16592)
Reported by: mwyres
Patches:
bug_16592.diff uploaded by snuffy (license 35)
Review: https://reviewboard.asterisk.org/r/539/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Three changes made here:
1) Do not fail if a previous log does not exist (in fact, this is probably
expected).
2) Ensure that the file descriptor to write to gets assigned properly. I am at
a loss as to why assigning safe_fd outside the if fixes this, but it makes
the if statement slightly less complicated anyway.
3) Move up the failure message so that the errno of the failure is not
overwritten by fclose.
(closes issue #16917)
Reported by: Artem
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@250565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r248859 | tilghman | 2010-02-25 15:21:05 -0600 (Thu, 25 Feb 2010) | 15 lines
Some platforms clear /var/run at boot, which makes connecting a remote console... difficult.
Previously, we only created the default /var/run/asterisk directory at install
time. While we could create it in the init script, that would not work for
those who start asterisk manually from the command line. So the safest thing
to do is to create it as part of the Asterisk boot process. This also changes
the ownership of the directory, because the pid and ctl files are created after
we setuid/setgid.
(closes issue #16802)
Reported by: Brian
Patches:
20100224__issue16802.diff.txt uploaded by tilghman (license 14)
Tested by: tzafrir
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@248861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@247335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
(closes issue #16642)
Reported by: kobaz
Patches:
groupmatch2.patch uploaded by kobaz (license 834)
Review: https://reviewboard.asterisk.org/r/503/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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!
(closes issue #15135)
Reported by: IgorG
Patches:
features.dynamic_park.v3.diff uploaded by IgorG (license 20)
2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7)
dynamic_parkinglot.diff uploaded by dvossel (license 671)
Tested by: eliel, IgorG, acunningham, mvanbaak, zktech
Review: https://reviewboard.asterisk.org/r/352/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247076 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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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
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
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
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
ao2_iterator_destroy was not being used on the
iterator during the test. This resulted in the
container never actually being destroyed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The options are: parkedcallparking, parkedcallhangup, parkedcallrecording, and
parkedcalltransfers. Previously these options were only available for the
default parking lot.
(closes issue #16641)
Reported by: bluecrow76
Patches:
asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76 (license 270)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When potentially sending manager events, return immediately if there are no
sessions or hooks. Also, avoid locking the hooks list if it is empty.
(issue #16455)
Reported by: atis
Patches:
manager_hooks_trunk.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In ast_frdup the frame data union does not get set to point to malloced memory
if the datalen is zero, so make sure to handle the same case in ast_frisolate
appropriately.
(closes issue #16058)
Reported by: atis
Patches:
bug16058-fix.patch uploaded by jpeeler (license 325)
Tested by: atis
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In order to improve readability, the output from 'test show
registered' has been modified to truncate fields to fit within
the format output if they are over a certain length.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@243200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/trunk@242919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When this code was developed, we came up with our own XML format for the test
output. I have since started looking at integration with other tools, namely
continuous integration frameworks, and this format seems to be supported
across a number of applications. With these changes in place, I was able
to get Atlassian Bamboo to interpret the test results.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Leif Madsen
Mark Michelson
Richard Mudgett
Russell Bryant
Tilghman Lesher
Kevin P. Fleming
Terry Wilson
Tzafrir Cohen
Jeff Peeler
Matthew Nicholson
Alec L Davis
David Vossel
Jason Parker
Sean Bright
Olle Johansson