The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change.
Patch by snuffy.
(closes issue #16559)
Reported by: cianmaher
Tested by: cianmaher, snuffy
Review: https://reviewboard.asterisk.org/r/461/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This directs users to documents which can help explain the
concepts and configuration options settable with the function.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258345 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows for multiple SRV queries to be done
from the dialplan for the same service on a single call while
still allowing one to bypass the call to SRVQUERY if they so
please.
Taking action since no comments had been left for a while.
This can easily be reverted if needed. External tests
still pass.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Change copyright date
* Place channel in autoservice when doing SRV lookup
* Get rid of trailing whitespace
* Change logic in load_module function
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
The PITCH_SHIFT function can be used on a channel to independently
modify the pitch of both rx and tx audio streams. Now you can
improve your conference calls by assigning a random pitch effect
to everyone entering a meetme room, or just make your day more
interesting by making your co-workers sound funny. These are just
some of the numerious practical uses for this function. Enjoy!
https://reviewboard.asterisk.org/r/526/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251038 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
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
Also include the tests provided by the reporter, as regression tests.
(closes issue #16667)
Reported by: wdoekes
Patches:
astsvn-func_match-off-by-one.diff uploaded by wdoekes (license 717)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When this patch was originally submitted, the code allowed for the token to be
set via a channel variable. I decided that a cleaner approach would be to
integrate it into the CHANNEL() function. Unfortunately, that is not a suitable
approach. It's not possible to get the value set on the channel soon enough
using that method. So, go back to the simple channel variable method.
(closes issue #16711)
Reported by: homesick
Patches:
iax-svn.diff uploaded by homesick (license 91)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243482 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This change makes the AES tests in test_substitution.c pass. We still need to
work through what's going wrong in the ast_str version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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: https://origsvn.digium.com/svn/asterisk/trunk@230994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Tilghman Lesher
Terry Wilson
Richard Mudgett
Bradley Latus
Mark Michelson
Russell Bryant
David Vossel
Jeff Peeler
Matthew Nicholson