Mark Michelson [Tue, 17 Sep 2013 16:09:46 +0000 (16:09 +0000)]
Switch transferdigittimeout to be configured as seconds instead of milliseconds.
This was an unintentional consequence of the update of features.conf to use the
config framework in Asterisk 12. Thanks to Marco Signorini on the Asterisk
developers list for pointing out the problem.
Kevin Harwell [Tue, 17 Sep 2013 14:48:06 +0000 (14:48 +0000)]
Confbridge: empty conference not being torn down
Confbridge would not properly tear down an empty conference bridge when all
users were kicked via end_marked=yes and at least one user was also set to
wait_marked. This occurred because while end_marked users were being kicked
and at least one was also set to wait_marked then the leave wait_marked handler
would be called on that user, but there would be no waiting user (still
considered active). The waiting users would decrement and now be negative. The
conference would remain, but be put into an inactive state. The solution was
to move from the active list to the wait list, those users with wait_marked set
right before kicking. This allows both the active and wait users to decrement
correctly and the confbridge to tear down properly.
A crashed also occurred when trying to list the specific conference from the CLI.
This happened because the conference specified was invalid. Since the
conference properly tears down now there is no way to reference it thus
alleviating the crash as well.
(closes issue ASTERISK-21859)
Reported by: Chris Gentle
Review: https://reviewboard.asterisk.org/r/2848/
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Richard Mudgett [Mon, 16 Sep 2013 18:34:55 +0000 (18:34 +0000)]
Fix module load errors for test_ari_model.so.
You cannot use a function pointer variable with an external function from
another dynamically loaded module because data variables are always
resolved even with RTLD_LAZY.
* Added wrapper functions for ast_ari_validate_int() and
ast_ari_validate_string() to use instead for the function pointer
variable.
(closes issue ASTERISK-22457)
Reported by: David M. Lee
Richard Mudgett [Mon, 16 Sep 2013 16:47:24 +0000 (16:47 +0000)]
chan_iax2: Fix saving the wrong expiry time in astdb.
When a new IAX2 client registers, the astdb database is updated with the
value of minregexpire defined in iax.conf instead of using the expiry time
that is provided by the client. The provided expiry time of the client is
updated after inserting the astdb entry. As a consequence, restarting or
reloading asterisk creates clients whose registration may expire before
they reregister. The clients are therefore unavailable after minregexpire
seconds until they reregister.
* Move updating of the expiry time to before inserting into the astdb.
(closes issue ASTERISK-22504)
Reported by: Stefan Wachtler
Patches:
chan_iax2.c.patch (license #6533) patch uploaded by Stefan Wachtler
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Matthew Jordan [Mon, 16 Sep 2013 02:33:23 +0000 (02:33 +0000)]
Filter internal channels out of bridge enter/leave message handling
Some channels exist merely as an implementation detail in Asterisk, such as
ConfBridge's announcer/recorder channels. These channels should never be
exposed to the outside world, or to interfaces that report on Asterisk. We
already filter out such channels in snapshot processing; however, we failed to
filter out bridge related messages that involved these channels.
This patch filters out bridge related messages that are for such channels. This
prevents a spurious WARNING message from being displayed when those channels
move in and out of bridges.
Richard Mudgett [Fri, 13 Sep 2013 22:05:07 +0000 (22:05 +0000)]
Restore Dial, Queue, and FollowMe 'I' option support.
The Dial, Queue, and FollowMe applications need to inhibit the bridging
initial connected line exchange in order to support the 'I' option.
* Replaced the pass_reference flag on ast_bridge_join() with a flags
parameter to pass other flags defined by enum ast_bridge_join_flags.
* Replaced the independent flag on ast_bridge_impart() with a flags
parameter to pass other flags defined by enum ast_bridge_impart_flags.
* Since the Dial, Queue, and FollowMe applications are now the only
callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the
calling contract to require the initial COLP exchange to already have been
done by the caller.
* Made all callers of ast_bridge_impart() check the return value. It is
important. As a precaution, I also made the compiler complain now if it
is not checked.
* Did some cleanup in parking_tests.c as a result of checking the
ast_bridge_impart() return value.
An independent, but associated change is:
* Reduce stack usage in ast_indicate_data() and add a dropping redundant
connected line verbose message.
David M. Lee [Fri, 13 Sep 2013 20:54:27 +0000 (20:54 +0000)]
Don't write to /tmp/refs when REF_DEBUG is not defined.
If MALLOC_DEBUG is enabled, then the debug destructor for the container
is used, which would erroneously write to /tmp/refs. This patch only
uses the debug destructor if ref_debug is used.
(closes issue ASTERISK-22536)
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David M. Lee [Fri, 13 Sep 2013 14:21:23 +0000 (14:21 +0000)]
res_pjsip: Forward PJSIP logging to Asterisk logging
This patch uses PJSIP's pj_log_set_log_func() to forward PJSIP's log
messages to Asterisk's logger. This is done in a new module:
res_pjsip_log_forwarder.so.
This patch sets defaultenabled on the existing res_pjsip_logger.so to
no, since logging every SIP packet seems a bit odd to do by default, and
is (hopefully) less necessary with regular PJSIP logging.
It also removes res_rtp_asterisk's disabling of PJSIP logging.
David M. Lee [Fri, 13 Sep 2013 14:18:36 +0000 (14:18 +0000)]
ARI: Fix WebSocket response when subprotocol isn't specified
When I moved the ARI WebSocket from /ws to /ari/events, I added code to
allow a WebSocket to connect without specifying the subprotocol if
there's only one subprotocol handler registered for the WebSocket.
Naively, I coded it to always respond with the subprotocol in use.
Unfortunately, according to RFC 6455, if the server's response includes
a subprotocol header field that "indicates the use of a subprotocol that
was not present in the client's handshake [...], the client MUST _Fail
the WebSocket Connection_.", emphasis theirs.
This patch correctly omits the Sec-WebSocket-Protocol if one is not
specified by the client.
Kinsey Moore [Fri, 13 Sep 2013 13:54:41 +0000 (13:54 +0000)]
Fix several crashes in MeetMeAdmin
This change ensures that MeetMeAdmin commands requiring a user actually
get a user and fixes another issue where an extra dereference could
occur for a last-entered user being ejected if a user identifier was
also provided.
(closes issue ASTERISK-21907)
Reported by: Alex Epshteyn
Review: https://reviewboard.asterisk.org/r/2844/
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Rusty Newton [Thu, 12 Sep 2013 23:21:35 +0000 (23:21 +0000)]
Documentation fix and improvements to XML configuration help res_pjsip_acl
* One bug fix. Made the synopsis for "type" to accurate.
* changing the usage of "IP-domains" to "IP addresses"
* clarifying the usage for the options, by adding a relevant description for
each
* modified other areas of the XML help for clarity, such as the module
description and a few synopsis changes here and there. See the patch.
Richard Mudgett [Thu, 12 Sep 2013 16:38:07 +0000 (16:38 +0000)]
core_local: Fix memory corruption race condition.
The masquerade super test is failing on v12 with high fence violations and
crashing. The fence violations are showing that party id allocated memory
strings are somehow getting corrupted in the
bridge_reconfigured_connected_line_update() function. The invalid string
values happen to be the freed memory fill pattern.
After much puzzling, I deduced that the
bridge_reconfigured_connected_line_update() is copying a string out of the
source channel's caller party id struct just as another thread is updating
it with a new value. The copying thread is using the old string pointer
being freed by the updating thread. A search of the code found the
unreal_colp_redirect_indicate() routine updating the caller party id's
without holding the channel lock.
A latent bug in v1.8 and v11 hatched in v12 because of the bridging and
connected line changes. :)
Rusty Newton [Thu, 12 Sep 2013 00:04:08 +0000 (00:04 +0000)]
'queue add member' help text correction
You are adding dial strings to the queue, not channels. An aribitrary string
could be used, but you are typically referencing a channel. Correcting the
command help text.
(issue ASTERISK-22263)
(closes issue ASTERISK-22263)
Reported By: Rusty Newton
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Jonathan Rose [Wed, 11 Sep 2013 19:56:48 +0000 (19:56 +0000)]
chan_sip: Reject calls without prior SDP on 200 OK
If we receive a 200 OK without SDP, we will now check to see if
the remote address has been established for that channel's RTP
session and if the to tag for that channel has changed from
the most recent to tag in a response less than 200.
If either a change has been made since the last to-tag was
received or the remote address is unset, then we will drop
the call.
(closes issue ASTERISK-22424)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/2827/diff/#index_header
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Kevin Harwell [Wed, 11 Sep 2013 14:14:03 +0000 (14:14 +0000)]
pjsip: reinvite for connected line updates occurs when it should not
Connected line updates are now only sent out if an actual update needs to occur.
This happens under the following conditions:
1. The endpoint we are sending to is trusted.
2. Either a P-Asserted-Identity or Remote Party-ID header needs to be added/sent.
3. The connected id's number and name are valid.
Also added an SDP when an update is sent out.
(closes issue AST-1212)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2831/
Richard Mudgett [Mon, 9 Sep 2013 23:23:02 +0000 (23:23 +0000)]
MALLOC_DEBUG: Change fence magic number to be completely different from the freed magic number.
Race conditions between freeing a nul terminated string and
ast_strdup()'ing it are more likely to be detected if the fence and freed
magic numbers are completely different.
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David M. Lee [Mon, 9 Sep 2013 20:12:46 +0000 (20:12 +0000)]
Fix DEBUG_THREADS when lock is acquired in __constructor__
This patch fixes some long-standing bugs in debug threads that were
exacerbated with recent Optional API work in Asterisk 12.
With debug threads enabled, on some systems, there's a lock ordering
problem between our mutex and glibc's mutex protecting its module list
(Ubuntu Lucid, glibc 2.11.1 in this instance). In one thread, the module
list will be locked before acquiring our mutex. In another thread, our
mutex will be locked before locking the module list (which happens in
the depths of calling backtrace()).
This patch fixes this issue by moving backtrace() calls outside of
critical sections that have the mutex acquired. The bigger change was to
reentrancy tracking for ast_cond_{timed,}wait, which wrongly assumed
that waiting on the mutex was equivalent to a single unlock (it actually
suspends all recursive locks on the mutex).
Matthew Jordan [Sun, 8 Sep 2013 23:25:55 +0000 (23:25 +0000)]
Update CDR Unit tests to reflect container changes in r398579
When a channel joins a multi-party bridge, the ordering of the CDRs that is
created is determined by the ordering of the channels who happen to be in that
bridge. When r398579 changed the number of buckets in the container to
something sensible, it changed the ordering that the CDRs was created in,
causing one of the multiparty tests to fail. This fixes the test with the
now expected ordering.
Kinsey Moore [Sat, 7 Sep 2013 01:02:04 +0000 (01:02 +0000)]
Prevent XMPP timeout on blank responses
Sometimes the Google Voice servers have a bad habit of sending out 1
byte replies to the xmpp resource. When a blank 1 byte reply is
received from the socket the buffer attempts to wait (endlessly) for
the rest of the reply from google which effectively blocks the socket
and google voice calls will no longer come into the server.
This patch allows the xmpp module to correctly detect empty packets and
send out ping replies to google. It also sets a socket timeout on the
default socket which prevents the xmpp socket from closing and
preventing future google voice calls from coming into the server.
Furthermore instead of sending an empty reply back to google we send a
proper xmpp ping reply back. This also adds several more
socket messages.
(closes issue ASTERISK-22347)
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/2771
Patches:
xmpp_fix_1.diff uploaded by Andrew Nagy (License #6524)
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The mailbox and context are swapped on the receiving end for all users
of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
versions. This swaps those values to be correct when publishing to the
internal event system from Jabber/XMPP distributed MWI state.
(closes issue ASTERISK-22435)
Reported by: abelbeck
Tested by: Michael Keuter
Patches:
asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
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Kinsey Moore [Fri, 6 Sep 2013 19:32:03 +0000 (19:32 +0000)]
Blocked revisions 398558
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Fix Jabber/XMPP distributed MWI
The mailbox and context are swapped on the receiving end for all users
of Jabber and XMPP distributed MWI in Asterisk 1.8 and all more recent
versions. This swaps those values to be correct when publishing to the
internal event system from Jabber/XMPP distributed MWI state.
(closes issue ASTERISK-22435)
Reported by: abelbeck
Tested by: Michael Keuter
Patches:
asterisk-1.8-res_jabber-aji_handle_pubsub_event.patch uploaded by abelbeck
asterisk-11-res_xmpp-xmpp_pubsub_handle_event.patch uploaded by abelbeck
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Kinsey Moore [Fri, 6 Sep 2013 18:49:09 +0000 (18:49 +0000)]
Fix build warnings
When AST_DEVMODE is not defined, ast_asserts are not compiled into the
binary. In some cases, this means variables are not referenced or are
set but unused which causes warnings to show up.
(closes issue ASTERISK-22446)
Reported by: Jason Parker (qwell)
Richard Mudgett [Thu, 5 Sep 2013 17:30:23 +0000 (17:30 +0000)]
chan_iax2: Fix stray reference to worker thread idle_list.
* Fix stray reference to idle_list in cleanup_thread_list(). This may be
the reason for the note in iax2_process_thread() about threads not being
removed from the task lists.
* Move cleanup_thread_list(&idle_list) to after the other lists are
cleaned up.
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Richard Mudgett [Thu, 5 Sep 2013 17:15:55 +0000 (17:15 +0000)]
chan_iax2: Fix bridgecallno deadlock avoidance.
* Fix bridgecallno deadlock avoidance. When doing deadlock avoidance, you
need to retest the status of values for each loop to see if you still need
the lock for bridgecallno.
* As a safety check, after acquiring the bridgecallno lock you should
check if iaxs[bridgecallno] is NULL just like the current callno checks.
* Move setting thread->iostate to IAX_IOSTATE_IDLE to after processing any
deferred frames to ensure that the iostate is IDLE when it is placed back
into the idle list. defer_full_frame() tries to ensure
iax2_process_thread() wakes up to process the frame.
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Richard Mudgett [Wed, 4 Sep 2013 16:00:00 +0000 (16:00 +0000)]
chan_misdn: Fix misdn debug output printed with arbitrary verbose levels.
Fix the misdn debug output to remote consoles. chan_misdn uses
ast_console_puts() which doesn't know about verbose levels. Better to use
ast_verbose() instead. Without this patch the misdn debug messages are
appended to the verbose level which ever was set by the message sent to
the console before, i.e. any undefined level.
David M. Lee [Fri, 30 Aug 2013 20:58:07 +0000 (20:58 +0000)]
Fix graceful shutdown crash.
The cleanup code for optional_api needs to happen after all of the optional
API users and providers have unused/unprovided. Unfortunately, regsitering the
atexit() handler at the beginning of main() isn't soon enough, since module
destructors run after that.
Kevin Harwell [Fri, 30 Aug 2013 19:51:48 +0000 (19:51 +0000)]
Add a reloadable option for sorcery type objects
Some configuration objects currently won't place nice if reloaded.
Specifically, in this case the pjsip transport objects. Now when
registering an object in sorcery one may specify that the object is
allowed to be reloaded or not. If the object is set to not reload
then upon reloading of the configuration the objects of that type
will not be reloaded. The initially loaded objects of that type
however will remain.
While the transport objects will not longer be reloaded it is still
possible for a user to configure an endpoint to an invalid transport.
A couple of log messages were added to help diagnose this problem if
it occurs.
(closes issue ASTERISK-22382)
Reported by: Rusty Newton
(closes issue ASTERISK-22384)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2807/
Matthew Jordan [Fri, 30 Aug 2013 18:35:21 +0000 (18:35 +0000)]
Update UPGRADE.txt file for Asterisk 12
This simply pulls in the changes that were breaking from the CHANGES file
and updates a few other areas accordingly. It also removes the 10 => 11
notes, which are traditionally removed from each major version and stored
in the appropriate UPGRADE-X.txt file.
Jonathan Rose [Fri, 30 Aug 2013 18:18:09 +0000 (18:18 +0000)]
features_config: Ignore parkinglots in features.conf instead of failing to load
Parkinglots are defined in res_features.conf now, but this patch fixes
features_config so that features don't fail to load when parkinglots
are present in features.conf
Richard Mudgett [Fri, 30 Aug 2013 17:02:13 +0000 (17:02 +0000)]
test_substituition: Fix failed test reporting to actually report failure.
You cannot put the "Testing <blah> pass/fail" on a single line before
actually performing the test. Now any additional failure information is
logged before the test pass/fail announcement.
* Added an additional CDR(answer,u) test.
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David M. Lee [Fri, 30 Aug 2013 13:39:35 +0000 (13:39 +0000)]
optional_api: Fix linking problems between modules that export global symbols
With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].
This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.
For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.
Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)
The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.
Other changes made as a part of this patch:
* The stubs for http_websocket that wrap system calls set errno to
ENOSYS.
* res_http_websocket now properly increments module use count.
* In loader.c, the while() wrappers around dlclose() were removed. The
while(!dlclose()) is actually an anti-pattern, which can lead to
infinite loops if the module you're attempting to unload exports a
symbol that was directly linked to.
* The special handling of nonoptreq on systems without weak symbol
support was removed, since we no longer rely on weak symbols for
optional_api.
[1]: https://wiki.asterisk.org/wiki/x/wACUAQ
(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
David M. Lee [Fri, 30 Aug 2013 13:17:12 +0000 (13:17 +0000)]
ARI: Implement /recordings/stored API's
his patch implements the ARI API's for stored recordings. While the
original task only specified deleting a recording, it was simple
enough to implement the GET for all recordings, and for an individual
recording.
The recording playback operation was modified to use the same code for
accessing the recording as the REST API, so that they will behave
consistently.
There were several problems with the api-docs that were also fixed,
bringing the ARI spec in line with the implementation. There were some
'wishful thinking' fields on the stored recording model (duration and
timestamp) that were removed, because I ended up not implementing a
metadata file to go along with the recording to store such information.
The GET /recordings/live operation was removed, since it's not really
that useful to get a list of all recordings that are currently going
on in the system. (At least, if we did that, we'd probably want to
also list all of the current playbacks. Which seems weird.)
Mark Michelson [Fri, 30 Aug 2013 00:10:49 +0000 (00:10 +0000)]
Sanitize XML output for PIDF bodies.
PJSIP's PIDF API does not replace angle brackets with
their appropriate counterparts for XML. So we have to
do it ourself. In this particular case, the problem had
to do with attempting to place an unsanitized SIP URI
into an XML node. Now we don't get a 488 from recipients
of our PIDF NOTIFYs.
Mark Michelson [Thu, 29 Aug 2013 22:53:17 +0000 (22:53 +0000)]
Fix method for creating activities string in PIDF bodies.
The previous method did not allocate enough space to create
the entire string, but adjusted the string's slen value to
be larger than the actual allocation. This resulted in garbled
text in NOTIFY requests from Asterisk.
This method allocates the proper amount of space first and then
writes the content into the buffer.
Kevin Harwell [Thu, 29 Aug 2013 22:45:15 +0000 (22:45 +0000)]
Verbose logging discrepancies
Refactored cases where a combination of ast_verbose/options_verbose were
present. Also in general tried to eliminate, in as many places as possible,
where the options_verbose global variable was being used. Refactored the way
local and remote consoles handle verbose message logging in an attempt to
solve the various discrepancies that sometimes would show between the two.
Mark Michelson [Thu, 29 Aug 2013 22:24:30 +0000 (22:24 +0000)]
Fix when the subscription_terminated callback is called for subscription handlers.
The previous placement would result in the resubscribe() callback called instead of
the subscription_terminated() callback being called when a subscription was ended
via a SUBSCRIBE request. This would result in confusing PJSIP and having it throw
an assertion.
Mark Michelson [Thu, 29 Aug 2013 21:33:19 +0000 (21:33 +0000)]
Fix a race condition where a canceled call was answered.
RFC 5407 section 3.1.2 details a scenario where a UAC sends
a CANCEL at the same time that a UAS sends a 200 OK for the
INVITE that the UAC is canceling. When this occurs, it is the
role of the UAC to immediately send a BYE to terminate
the call.
This scenario was reproducible by have a Digium phone with two lines
place a call to a second phone that forwarded the call to the second
line on the original phone. The Digium phone, upon realizing that it
was connecting to itself, would attempt to cancel the call. The timing
of this happened to trigger the aforementioned race condition about
80% of the time. Asterisk was not doing its job of sending a BYE
when receiving a 200 OK on a cancelled INVITE. The result was that
the ast_channel structure was destroyed but the underlying SIP
session, as well as the PJSIP inv_session and dialog, were still
alive. Attempting to perform an action such as a transfer, once in
this state, would result in Asterisk crashing.
The circumstances are now detected properly and the session is ended
as recommended in RFC 5407.
Matthew Jordan [Thu, 29 Aug 2013 20:21:42 +0000 (20:21 +0000)]
Revert r394939 due to (numerous) objections
The patch from ASTERISK-21965 was committed perhaps a bit too hastily. Walter
and Tzafrir have pointed out numerous issues with the approach and have
propsed an alternative in r/2757. Since it's not a time critical issue and
is not worth holding up the release of 12 for it, I've gone ahead and reverted
r394939 from 12/trunk and re-opened ASTERISK-21965.
Matthew Jordan [Thu, 29 Aug 2013 16:04:42 +0000 (16:04 +0000)]
Recursively search for '.c' files when making documentation with 'make full'
Without this, documentation defined in sub-folders is ignored. Since having
properly generated documentation is especially important in Asterisk 12 -
not having it can cause a module to not load - 'make full' needs to look in
all .c files.
Richard Mudgett [Wed, 28 Aug 2013 23:14:57 +0000 (23:14 +0000)]
Fix some uninitialized buffers for CDR handling valgrind found.
* Made ast_strftime_locale() ensure that the output buffer is initialized.
The std library strftime() returns 0 and does not touch the buffer if it
has an error. However, the function can also return 0 without an error.
Richard Mudgett [Wed, 28 Aug 2013 22:55:27 +0000 (22:55 +0000)]
Fixed problems with ast_cdr_serialize_variables().
* Fixed return value of ast_cdr_serialize_variables() on error. It needs
to return 0 indicating no CDR variables found.
* Made ast_cdr_serialize_variables() check the return value of
cdr_object_format_property() and assert if nonzero. A member of the
cdr_readonly_vars[] was not handled.
* Removed unused elements from cdr_readonly_vars[]: total_duration,
total_billsec, first_start, and first_answer.
Mark Michelson [Wed, 28 Aug 2013 21:05:31 +0000 (21:05 +0000)]
Improve detection of answer on SIP blind transfer.
A problem encountered during testing was that res_pjsip_refer would
not ever send a NOTIFY with a 200 OK sipfrag. This is because the framehook
that was supposed to send the NOTIFY would never be told that an answer
had occurred. This happened for two reasons:
1) The transferee channel on which the framehook was on was already up.
2) Answers are rarely if ever written to channels. Rather, the ast_answer()
or ast_raw_answer() function is used to answer channels.
Thanks to a suggestion by Matt Jordan, the best way to detect that the call
had been answered was to find out when the transferee channel joined a bridge.
With stasis this is an easy task. So now, in addition to the framehook logic,
there is a stasis subscription used to determine when the transferee has entered
a bridge. Once it has entered, an appropriate NOTIFY is sent.
Matthew Jordan [Wed, 28 Aug 2013 20:55:31 +0000 (20:55 +0000)]
Add database schema management using Alembic
This patch replaces contrib/realtime/ with a new setup for managing the
database schema required for database integration with Asterisk. In
addition to initializing a database with the proper schema, alembic can do a
database migration to assist with upgrading Asterisk in the future.
Hopefully this helps make setting up and operating Asterisk with a database
easier.
With this the schema only needs to be maintained in one place instead of
once per database. The schemas I have added here have a bit of improvement
over the examples that were there before (some added consistency and added
some missing indexes). Managing the schema in one place here also applies
to all databases supported by SQLAlchemy.
See contrib/ast-db-manage/README.md for more details.
projects / thirdparty / asterisk.git / log
