chan_pjsip: Don't change formats when frame of unsupported format is received.
Receipt of an RTP packet currently causes the formats on an PJSIP channel to
change to the format of the RTP packet. In some off-nominal cases it's possible
for this to be a format that has not been configured or negotiated. This change
makes it so only formats explicitly configured on the endpoint are allowed.
Matt Jordan [Wed, 8 Jul 2015 21:38:24 +0000 (16:38 -0500)]
res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails
Having a debug message tell us that we attempted to look up an item but
failed is nice in circumstances when it isn't clear if the wizard was
queried correctly or not.
Matt Jordan [Wed, 8 Jul 2015 21:35:36 +0000 (16:35 -0500)]
res_pjsip/configuration: Fix a variety of default value problems
This patch fixes some bad default value handling in the following
settings:
* The 'message_context' and 'accountcode' settings are not mandatory. As
such, we can allow their stringfield values to be empty.
* The 'media_encryption' setting applies a default value of 'none' to
the setting, which it then can't parse or understand. Since the value
is documented to be 'no', this will now apply that as the default
value.
Matt Jordan [Wed, 8 Jul 2015 21:32:25 +0000 (16:32 -0500)]
main/sorcery: Provide log messages when a wizard does not support an operation
If a sorcery wizard does not support one of the 'optional' CRUD
operations (namely the CUD), log a WARNING message so we are aware of
why the operation failed. This also removes an assert in this case, as
the CUD operation may have been triggered by an external system, in
which case it is not a programming error but a configuration error.
Richard Mudgett [Fri, 10 Jul 2015 23:17:52 +0000 (18:17 -0500)]
res_pjsip_session.c: Fix crash on call disconnect.
The crash fix for ASTERISK-25183 backported some code from master to try
to make sure that a BYE response is processed by the same serializer used
by the BYE request. The identified race condition causing that backport
was the BYE request code had not finished processing after sending the BYE
before the BYE response came in for processing under a different thread.
Unfortunately, there is still a race condition. Now the race condition is
between destroying the call session's serializer in
ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a
reference to the serializer for a BYE response. Even worse, the new race
condition is a design limitation of the taskprocessor implementation that
didn't matter in versions before v12. Back then, taskprocessors were only
destroyed when a module unloaded. Now res_pjsip can destroy them when a
call ends.
However, as noted on the ASTERISK-25183 commit,
session_inv_on_state_changed() is disassociating the dialog from the
session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED.
This is a tad too soon because our BYE request transaction has not
completed yet.
* Split session_end() that is called by session_inv_on_state_changed() to
hold off session destruction until the BYE transaction timeout occurs or a
failed initial INVITE transaction timeout occurs in
session_inv_on_tsx_state_changed().
Benjamin Ford [Tue, 14 Jul 2015 18:12:32 +0000 (13:12 -0500)]
ARI: Added new functionality to reload a single module.
An http request can be sent to reload an Asterisk module. If the
module can not be reloaded or is not already loaded, an error
response will be returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, based
on configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be reloaded through http requests
Benjamin Ford [Tue, 14 Jul 2015 13:55:14 +0000 (08:55 -0500)]
ARI: Added new functionality to unload a single module.
An http request can be sent to unload an Asterisk module. If the
module can not be unloaded or is already unloaded, an error response
will be returned.
The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, depending
on configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be unloaded through http requests
Benjamin Ford [Mon, 13 Jul 2015 21:00:19 +0000 (16:00 -0500)]
ARI: Added new functionality to load a single module.
An http request can be sent to load an Asterisk module. If the
module can not be loaded or is loaded already, an error response
will be returned.
The command curl -v -u user:pass -X POST 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending on
configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be loaded through http requests
Benjamin Ford [Mon, 13 Jul 2015 15:54:51 +0000 (10:54 -0500)]
ARI: Added new functionality to get information on a single module.
An http request can be sent to retrieve information on a single
module, including the resource name, description, use count, status,
and support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending on
configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on a single module can now be retrieved
Kevin Harwell [Wed, 8 Jul 2015 19:56:10 +0000 (14:56 -0500)]
bridge.c: Fixed race condition during attended transfer
During an attended transfer a thread is started that handles imparting the
bridge channel. From the start of the thread to when the bridge channel is
ready exists a gap that can potentially cause problems (for instance, the
channel being swapped is hung up before the replacement channel enters the
bridge thus stopping the transfer). This patch adds a condition that waits
for the impart thread to get to a point of acceptable readiness before
allowing the initiating thread to continue.
Matt Jordan [Wed, 8 Jul 2015 21:28:13 +0000 (16:28 -0500)]
main/sorcery: Don't fail object set creation from JSON if field fails
Some individual fields may fail their conversion due to their default
values being invalid for their custom handlers. In particular,
configuration values that depend on others being enabled (and thus have
an empty default value) are notorious for tripping this routine up. An
example of this are any of the DTLS options for endpoints. Any of the
DTLS options will fail to be applied (as DTLS is not enabled), causing
the entire object set to be aborted.
This patch makes it so that we log a debug message when skipping a
field, and rumble on anyway.
Matt Jordan [Wed, 8 Jul 2015 21:21:09 +0000 (16:21 -0500)]
main/format_cap: Parse capabilities generated by ast_format_cap_get_names
We have a strange relationship between the parsing of format
capabilities from a string and their representation as a string. We
expect the format capabilities to be expressed as a string in the
following format:
allow = !all,ulaw,alaw
disallow = g722
While we would generate the string representation of those formats as:
When the configuration framework needs to store values as a string, it
generates the format capabilities using the second representation; this
representation however cannot be parsed when the entry is rehydrated.
This patch fixes that by updating
ast_format_cap_update_by_allow_disallow to parse an entry as if it were
in the generated format if it has a leading '(' and a trailing ')'.
Matt Jordan [Sat, 27 Jun 2015 22:53:37 +0000 (17:53 -0500)]
tests/test_devicestate: Add additional tests for the device state API
This patch adds more tests that exercise the device state API. This includes:
* Tests that cover adding a device state provider, as well as deleting a
device state provider. This also verifies that you cannot add an
already added device state provider, and cannot delete an already
deleted device state provider.
* A test that covers changing device state and receiving said updates
from a device state subscriber. This also covers hitting both the
device state cache as well as a custom device state provider.
* A test that covers converting device state to channel state and device
state values to a string representation and back.
* A test that covers obtaining device state from an active channel and a
channel driver that provides its own device state.
Matt Jordan [Sat, 27 Jun 2015 22:51:43 +0000 (17:51 -0500)]
main/devicestate: Prevent duplicate registration of device state providers
Currently, the device state provider API will allow you to register a
device state provider with the same case insensitive name more than
once. This could cause strange issues, as the duplicate device state
providers will not be queried when a device's state has to be polled.
This patch updates the API such that a device state provider with the
same name as one that has already registered will be rejected.
Matt Jordan [Sat, 11 Jul 2015 03:25:00 +0000 (22:25 -0500)]
res/res_sorcery_memory_cache: Fix test registration issues
Again, tests now need to not end with a newline. This patch makes it so
the tests can register again, unit tests will actually pass, and we can
stop wasting time trying to figure out why builds are failing when they
really aren't failing.
Matt Jordan [Sat, 11 Jul 2015 02:42:58 +0000 (21:42 -0500)]
tests/test_sorcery_memory_cache_thrash: Fix test loading problems
Because unit tests now want descriptions to not end with a newline, the
sorcery memory cache thrash tests failed to register. This patch
corrects their descriptions.
Benjamin Ford [Fri, 26 Jun 2015 15:57:15 +0000 (10:57 -0500)]
ARI: Added new functionality to get all module information.
An http request can be sent to retrieve a list of all existing modules,
including the resource name, description, use count, status, and
support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/
asterisk/modules" (or something similar, depending on configuration)
can be run in the terminal to access this new functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on modules can now be retrieved
bridge_native_rtp.c: Don't start native RTP bridging after attended transfer.
The bridge_native_rtp module adds a frame hook to channels which are in
a native RTP bridge. This frame hook is used to intercept when a hold
or unhold frame traverses the bridge so native RTP can be stopped or
started as appropriate. This is expected but exposes a specific bug
when attended transfers are involved.
Upon completion of an attended transfer an unhold frame is queued up
to take one of the channels involved off hold. After this is done
the channel is moved between bridges.
When the frame hook is involved in this case for the unhold it
releases the channel lock and acquires the bridge lock. This
allows the bridge core to step in and move the channel
(potentially changing the bridging techology) from another thread.
Once completed the bridge lock is released by the bridge core.
The frame hook is then able to acquire the bridge lock and
wrongfully starts native RTP again, despite the channel no longer
being in the bridge or needing to start native RTP. In fact at
this point the frame hook is no longer attached to the channel.
This change makes it so the native RTP bridge data is available to
the frame hook when it is invoked. Whether the frame hook has
been detached or not is stored on the native RTP bridge data and
is checked by the frame hook before starting or stopping native
RTP bridging. If the frame hook has been detached it does nothing.
Joshua Colp [Sat, 16 May 2015 22:02:50 +0000 (19:02 -0300)]
res_sorcery_memory_cache: Backport to 13
Gerrit is complaining of conflicts when trying to create a patch series
of all of the cherry-picked master commits, so I have instead squashed
it all into one commit.
res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.
This change moves logic for setting up the DTLS SSL contexts to
when the SDP is done being processed instead of when ICE negotiation
completes. It also stops handshakes from being initiated when we
are acting as a server.
Manipulating the SSL context when ICE negotiation has completed
is problematic as the SSL context is not protected and if acting
as a client the remote side may have started DTLS negotiation
already.
The retransmission timeout timer code has also been split up
and simplified some. Both RTP and RTCP now have their own timers
and the points at which the timer is stopped and started is now
more specific. When a packet is sent the timer is started. When
a response is received but before it is processed the timer is
stopped. This provides a guarantee that the timeout is not
occurring while the response is processed.
MWI subscriptions can crash or corrupt memory when using the subscription
datastore to access the MWI subscription object because the datastore is
not holding a reference to the object.
* Give the subscription datastore a ref to the MWI subscription object.
It is unfortunate that the ref causes a circular ref chain that must be
explicitly broken to allow the memory to get released. The loop is broken
when the subscription is shutdown and if the subscription setup fails.
When res_pjsip body generator modules were generating XML or XPIDF
response bodies, there was a chance that the generated body would be the
exact size of the supplied buffer. Adding the nul string terminator would
then write beyond the end of the buffer and potentially corrupt memory.
* Fix MALLOC_DEBUG high fence violations caused by adding a nul string
terminator on the end of a buffer for XML or XPIDF response bodies.
* Made calls to pj_xml_print() safer if the XML prolog is requested. Due
to a bug in pjproject, the return value could be -1 _or_
AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough.
* Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the
return value of pj_xml_print() when the supplied buffer is not large
enough.
When a caller calls a FAX number and then hangs up right after the call is
answered then the T.38 re-INVITE automatic reject timer may still be
running after the channel goes away.
* Added session NULL channel checks on the code paths that get executed by
t38_automatic_reject() to prevent a crash when the T.38 re-INVITE
automatic reject timer expires.
Richard Mudgett [Fri, 5 Jun 2015 20:37:33 +0000 (15:37 -0500)]
res_pjsip: Need to use the same serializer for a pjproject SIP transaction.
All send/receive processing for a SIP transaction needs to be done under
the same threadpool serializer to prevent reentrancy problems inside
pjproject and res_pjsip.
* Add threadpool API call to get the current serializer associated with
the worker thread.
* Pick a serializer from a pool of default serializers if the caller of
res_pjsip.c:ast_sip_push_task() does not provide one.
This is a simple way to ensure that all outgoing SIP request messages are
processed under a serializer. Otherwise, any place where a pushed task is
done that would result in an outgoing out-of-dialog request would need to
be modified to supply a serializer. Serializers from the default
serializer pool are picked in a round robin sequence for simplicity.
A side effect is that the default serializer pool will limit the growth of
the thread pool from random tasks. This is not necessarily a bad thing.
* Made pjsip_distributor.c save the thread's serializer name on the
outgoing request tdata struct so the response can be processed under the
same serializer.
NOTE: session_inv_on_state_changed() is disassociating the dialog from the
session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED.
Unfortunately this is a tad too soon because our BYE request transaction
has not completed yet.
This change makes it so that when accepting a WebSocket
connection the HTTP response is sent as one packet instead of
fragmented. Browsers don't like it when you send it fragmented.
Matt Jordan [Sat, 27 Jun 2015 23:47:19 +0000 (18:47 -0500)]
Makefile: Remove coverage files on 'make clean'
This patch updates a variety of Makefiles in Asterisk's build system to
remove .gcda and .gcno files when 'make clean' is executed. These files
are generated when '--enable-coverage' is passed to the Asterisk
configure script.
Walter Doekes [Thu, 2 Jul 2015 14:08:12 +0000 (16:08 +0200)]
chan_sip: Fix early call pickup channel leak.
When handle_invite_replaces() was called, and either ast_bridge_impart()
failed or there was no bridge (because the channel we're picking up was
still ringing), chan_sip would leak a channel.
Thanks Matt and Corey for checking the bridge path.
Walter Doekes [Thu, 2 Jul 2015 11:10:59 +0000 (13:10 +0200)]
rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format.
When running valgrind on Asterisk, it complained about:
==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304)
==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...)
==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292)
==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437)
The code in question is a struct assignment, which may be performed by
memcpy as a compiler optimization. It is changed to only copy the struct
contents if source and destination are different.
Walter Doekes [Thu, 2 Jul 2015 10:16:26 +0000 (12:16 +0200)]
astfd: Fix buffer overflow in DEBUG_FD_LEAKS.
If DEBUG_FD_LEAKS was used and more file descriptors than the default of
1024 were available, some DEBUG_FD_LEAKS-patched functions would
overwrite memory past the fixed-size (1024) fdleaks buffer.
This change:
- adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe
- consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024
- stores pointers to constants instead of copying the contents
- reorders the fdleaks struct for possibly tighter packing
- adds a tiny bit of documentation
Walter Doekes [Thu, 2 Jul 2015 09:57:44 +0000 (11:57 +0200)]
res_timing: Don't close FD 0 when out of open files.
This fixes so a failure to get a timer file descriptor does not cascade
to closing FD 0.
On error, both res_timing_kqueue and res_timing_timerfd would call the
destructor before setting the file handle. The file handle had been
initialized to 0, causing FD 0 to be closed. This in turn, resulted in
floods of "CLI>" messages and an unusable terminal.
ASTERISK-19277 #close
Reported by: Barry Chern
For the 13 branch, this was already fixed. This patch only ensures that
we do not attempt to close a negative file descriptor.
Matt Jordan [Wed, 1 Jul 2015 21:04:50 +0000 (16:04 -0500)]
sorcery/realtime: Add a bit of debug and warning messages for bad configs
When a mapping does not exist between a sorcery.conf defined object and
a realtime mapping in extconf, currently, the user will receive a slew
of ERROR messages that don't really tell what is happening. Some ERROR
messages may even be misleading, as they occur after the sorcery API has
already given up on the attempt to load and create the sorcery object.
This patch adds a bit of debug and a useful WARNING message for when a
wizard's open callback fails for a particular object type. In the bad
configurations that resulted in this patch, this provided a 'root cause'
WARNING message that pointed in the right direction of the configuration
problem.
Mark Michelson [Mon, 29 Jun 2015 17:45:02 +0000 (12:45 -0500)]
res_sorcery_realtime: Fix leak of sorcery object type.
This prevents a leak of a sorcery object type when realtime sorcery
objects are retrieved by fields or when multiple objects are retrieved.
The extent of this leak is that sorcery object types would be leaked.
These are allocated whenever an object type is registered with sorcery,
meaning that on module shutdown, these objects would be leaked. This
could be problematic if many reloads were performed, but it is not as
severe as if every sorcery object retrieved from realtime were being
leaked.
Matt Jordan [Sat, 27 Jun 2015 03:02:42 +0000 (22:02 -0500)]
res/res_corosync: Always decline module load, instead of failing
Returns a 'failure' from the module load routine indicates to Asterisk
that it should abort loading completely. This is rarely - in fact,
really, never - a good option. Aborting load of Asterisk from a dynamic
module implies that the core, and the rest of the dynamic modules, don't
matter: we should abandon all processing.
res_corosync is really not that important.
This patch updates the module such that, if it fails to load, it
politely declines (emitting ERROR messages along the way), and allows
Asterisk to continue to function.
Note that this issue was keeping Asterisk unit tests from running on
certain build agents.
Matt Jordan [Sat, 27 Jun 2015 01:38:58 +0000 (20:38 -0500)]
main/pbx: Resolve case sensitivity regression in PBX hints
When 8297136f was merged for ASTERISK-25040, a regression was introduced
surrounding the case sensitivity of device names within hints.
Previously, device names - such as 'sip/foo' - were compared in a case
insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After
that patch, only the case sensitive name would match, i.e., 'SIP/foo'.
As a result, some dialplan hints stopped working.
This patch re-introduces case insensitive matching for device names in
hints.
Mark Michelson [Fri, 26 Jun 2015 21:12:33 +0000 (16:12 -0500)]
res_pjsip_nat: Adjust when contact should be rewritten.
A previous change made the contact only get rewritten if the dialog's
route set was not marked frozen. Unfortunately, while the intent of this
is correct, the dialog's route set actually gets marked as frozen
earlier than expected, especially for UAS dialogs.
Instead, the idea is that the contact needs to not be rewritten if there
is a pre-existing route set on the dialog. This is now accomplished by
checking the dialog's route set list instead of checking if the route
set is frozen.
Doing this causes some broken tests to begin passing again.
Richard Mudgett [Fri, 19 Jun 2015 23:27:24 +0000 (18:27 -0500)]
res_pjsip_outbound_registration.c: Add a serializer shutdown group.
The client_state objects contain a serializer used to send the outbound
REGISTER messages. Once all those message transactions are complete then
the module can shutdown.
res_pjsip_refer will attempt to add Referred-By or Replaces headers to
outbound INVITEs at times. If the INVITE gets challenged for
authentication, then we will resend the INVITE. Prior to this patch, the
Referred-By or Replaces header would be re-added to the outbound INVITE,
resulting in duplicated headers.
Mark Michelson [Tue, 23 Jun 2015 22:43:31 +0000 (17:43 -0500)]
res_pjsip_nat: Rewrite route set when required.
When performing some provider testing, the rewrite_contact option was
interfering with proper construction of a route set when sending an ACK
after receiving a 200 OK response to an INVITE.
The initial INVITE was sent to address sip:foo. The 200 OK had a Contact
header with URI sip:bar. In addition, the 200 OK had Record-Route
headers for sip:baz and sip:foo, in that order. Since the Record-Route
headers had the lr parameter, the result should have been:
* Set R-URI of the ACK to sip:bar.
* Add Route headers for sip:foo and sip:baz, in that order.
However, the rewrite_contact option resulted in our rewriting the
Contact header on the 200 OK to sip:foo. The result was:
* R-URI remained sip:foo.
* We added Route headers for sip:foo and sip:baz, in that order.
The result was that sip:bar was not indicated in the ACK at all, so the
far end never received our ACK. The call eventually dropped.
The intention of rewrite_contact is to rewrite the most immediate
destination of our SIP request to be the same address on which we
received a request or response. In the case of processing a SIP response
with Record-Route headers, this means that instead of rewriting the
Contact header, we should instead rewrite the bottom-most Record-Route
header. In the case of processing a SIP request with Record-Route
headers, this means we rewrite the top-most Record-route header.
Like when we rewrite the Contact header, we also ensure to update
the dialog's route set if it exists.
* handle_client_state_destruction() must always be passed a ref to
client_state because it will always unref client_state.
handle_registration_response() was not passing a client_state ref.
* Made the final un-REGISTER message get sent normally using the pjproject
register control structure in handle_client_state_destruction(). The
previous code attempted to short circuit the response handling for the
module to unload. That doesn't work for a couple reasons. One,
pjsip_regc_send() may call the registered callback before it returns and
unbalance the client_state ref count. Two, the registered callback
handles any authentication for the un-REGISTER message.
* Made the distinction between internal registration state and external
registration status with sip_outbound_registration_status_str(). This is
necessary to avoid altering documented AMI messages with internal
changes.
* Removed references to client_state->client outside of the serializer
thread. When handle_client_state_destruction() destroys the pjproject
register control structure that memory is freed and cannot be referenced
anymore. These accesses were to provide information for debug and
off-nominal warning messages.
* In sip_outbound_registration_timer_cb() you should not access entry->id
after unrefing client_state because the passed in entry is normally
pointing to the timer entry in the client_state object.
Richard Mudgett [Mon, 15 Jun 2015 20:28:41 +0000 (15:28 -0500)]
res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API
The sorcery pjsip 'registration' config object needs to be destroyed on
module unload. Otherwise, a reload of res_pjsip could try to use
callbacks for a previously unloaded instance of the module provided by
ast_sorcery_object_register() or one of the variants. Also, if
res_pjsip_outbound_registration were subsequently reloaded, the sorcery
config field objects would be registered in sorcery twice.
Joshua Colp [Thu, 25 Jun 2015 11:42:46 +0000 (08:42 -0300)]
channel: Remove ignore of answer on non-outgoing channels.
Due to the way that channels can now be moved around inside of
Asterisk it is possible for the outgoing flag of a channel to get
cleared before it has been answered. This results in the bridge
not receiving notification that the outgoing leg has been answered.
This most easily exhibits itself with DTMF based blond transfers.
Since the answer of the outgoing leg is ignored the other party
continues to receive both a locally generated ringing and the
media stream of the outgoing leg upon its answer. This results
in no media being heard.
This change removes the ignore of the answer and allows it
to pass through.
Richard Mudgett [Mon, 15 Jun 2015 20:28:00 +0000 (15:28 -0500)]
sorcery: Add ast_sorcery_object_unregister() API call.
Find and unlink the specified sorcery object type to complement
ast_sorcery_object_register(). Without this function you cannot
completely unload individual modules that use sorcery for configuration.
Richard Mudgett [Mon, 15 Jun 2015 18:38:58 +0000 (13:38 -0500)]
res_pjsip_outbound_registration.c: Reorder load_module() and unload_module().
It is best if the loading code creates and initializes the module's
infrastructure before letting the system know of its existence. The
unloading code needs to reverse the actions of the loading code and in the
reverse order.
Richard Mudgett [Wed, 24 Jun 2015 19:39:01 +0000 (14:39 -0500)]
Unit tests: Fix unit test description strings.
Analyzing the code shows that the unit test summary and description
strings should not end with a new-line character. Where these strings are
used in the code a new-line is provided for output.
Joshua Colp [Tue, 23 Jun 2015 16:21:41 +0000 (13:21 -0300)]
app_dial: Hold reference to calling channel formats when dialing outbound.
Currently when requesting a channel the native formats of the
calling channel are provided to the core for usage when dialing
the outbound channel. This occurs without holding the channel lock
or keeping a reference to the formats. This is problematic as
the channel driver may end up changing the formats during this time.
In the case of chan_sip this happens when an SDP negotiation
completes.
This change makes it so app_dial keeps a reference to the native
formats of the calling channel which guarantees that they will
remain valid for the period of time needed.
projects / thirdparty / asterisk.git / log
