Implements the following cli commands:
pjsip list aors
pjsip list auths
pjsip list channels
pjsip list contacts
pjsip list endpoints
pjsip show aor(s)
pjsip show auth(s)
pjsip show channels
pjsip show endpoint(s)
Also...
Minor modifications made to the AMI command implementations to facilitate
reuse.
New function ast_variable_list_sort added to config.c and config.h to implement
variable list sorting.
(issue ASTERISK-22610)
patches:
pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
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AMI has received substantial updates over the past year. Not only has the
syntax been vastly improved and made consistent (which entails many event
changes), but the underlying things that those events convey have changed
substantially as well.
After some conversation in #asterisk-dev, it was agreed that this is a good
time to jump to 2.
At the same time, since ARI will most likely use semantic versioning, we
might as well use that for AMI as well. That also affords us greater meaning
for the AMI version.
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For the explanation, here is a copy-paste of the review board explanation:
Initially, it was discovered that performing an attended transfer of a
multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread
started a masquerade and reached the point where it was calling the fixup()
callback on the "original" channel. For chan_pjsip, this involves pushing a
synchronous task to the session's serializer. The problem was that a task ahead
of the fixup task was also attempting to perform a channel masquerade. However,
since masquerades are designed in a way to only allow for one to occur at a
time, the task ahead of the fixup could not continue until the masquerade
already in progress had completed. And of course, the masquerade in progress
could not complete until the task ahead of the fixup task had completed.
Deadlock.
The initial fix was to change the fixup task to be asynchronous. While this
prevented the deadlock from occurring, it had the frightful side effect of
potentially allowing for tasks in the session's serializer to operate on a
zombie channel.
Taking a step back from this particular deadlock, it became clear that the
problem was not really this one particular issue but that masquerades
themselves needed to be addressed. A PJSIP attended transfer operation calls
ast_channel_move(), which attempts to both set up and execute a masquerade. The
problem was that after it had set up the masquerade, the PBX thread had swooped
in and tried to actually perform the masquerade. Looking at changes that had
been made to Asterisk 12, it became clear that there never is any time now that
anyone ever wants to set up a masquerade and allow for the channel thread to
actually perform the masquerade. Everyone always is calling ast_channel_move(),
performs the masquerade itself before returning.
In this patch, I have removed all blocks of code from channel.c that will
attempt to perform a masquerade if ast_channel_masq() returns true. Now, there
is no distinction between setting up a masquerade and performing the
masquerade. It is one operation. The only remaining checks for
ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not
want to interrupt a masquerade by hanging up the channel. Instead, now
ast_hangup() will wait for a masquerade to complete before moving forward with
its operation.
The ast_channel_move() function has been modified to basically in-line the
logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has
been killed off for real. ast_channel_move() now has a lock associated with it
that is used to prevent any simultaneous moves from occurring at once. This
means there is no need to make sure that ast_channel_masq() or
ast_channel_masqr() are already set on a channel when ast_channel_move() is
called. It also means the channel container lock is not pulling double duty by
both keeping the container locked and preventing multiple masquerades from
occurring simultaneously.
The ast_do_masquerade() function has been renamed to do_channel_masquerade()
and is now internal to channel.c. The function now takes explicit arguments of
which channels are involved in the masquerade instead of a single channel.
While it probably is possible to do some further refactoring of this method, I
feel that I would be treading dangerously. Instead, all I did was change some
comments that no longer are true after this changeset.
The other more minor change introduced in this patch is to res_pjsip.c to make
ast_sip_push_task_synchronous() run the task in-place if we are already a SIP
servant thread. This is related to this patch because even when we isolate the
channel masquerade to only running in the SIP servant thread, we would still
deadlock when the fixup() callback is reached since we would essentially be
waiting forever for ourselves to finish before actually running the fixup. This
makes it so the fixup is run without having to push a task into a serializer at
all.
(closes issue ASTERISK-22936)
Reported by Jonathan Rose
Review: https://reviewboard.asterisk.org/r/3069
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This change is in preparation for external MWI support.
Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context. The
only exception is the legacy hasvoicemail users.conf option. The legacy
option will only work for app_voicemail mailboxes. The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.
chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier. chan_sip just stored and
compared the two components. chan_dahdi actually used the box
information.
The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number. As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.
Review: https://reviewboard.asterisk.org/r/3072/
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When doing the rework of the CDR engine that pushed all of the logic into cdr.c
and made it respond to changes in channel state over Stasis, we knew that
accessing the CDR engine from the dialplan would be "slightly"
non-deterministic. Dialplan threads would be accessing CDRs while Stasis
threads would be updating the state of said CDRs - whereas in the past,
everything happened on the dialplan threads. Tests have shown that "slightly"
is in reality "very".
This patch synchronizes things by making the dialplan applications/functions
that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and
CDR_PROP now all use Stasis to send their requests over to the CDR engine,
and synchronize on the channel Stasis topic via a subscription so that they
return their values/control to the dialplan at the appropriate time.
While going through this, the following changes were also made:
* DISA, which can reset the CDR when a user successfully authenticates, now
just uses the ResetCDR app to do this. This prevents having to duplicate
the same Stasis synchronization logic in that application.
* Answer no longer disables CDRs. It actually didn't work anyway - calling
DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer
time - it just kills all CDRs on that channel, which isn't what the caller
would intend.
(closes issue ASTERISK-22884)
(closes issue ASTERISK-22886)
Review: https://reviewboard.asterisk.org/r/3057/
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Original commit message by mmichelson (asterisk 12 r403311):
"This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such."
The above was initially committed and then reverted at r403398. The problem
was found to be in core_local.c in the publish_local_bridge_message function.
The ast_unreal_lock_all function locks and adds a reference to the returned
channels and while they were being unlocked they were not being unreffed when
no longer needed. Fixed by unreffing the channels.
Also in bridge.c a lock was obtained on "other->chan", but then an attempt was
made to unlock "other" and not the previously locked channel. Fixed by
unlocking "other->chan"
(closes issue ASTERISK-22709)
Reported by: John Bigelow
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Bridges have two new optional properties, a creator and a name.
Certain consumers of bridges will automatically provide bridges that
they create with these properties. Examples include app_bridgewait,
res_parking, app_confbridge, and app_agent_pool. In addition, a name
may now be provided as an argument to the POST function for creating
new bridges via ARI.
(closes issue AFS-47)
Review: https://reviewboard.asterisk.org/r/3070/
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This patch allows individual dialplan functions to be marked as
'dangerous', to inhibit their execution from external sources.
A 'dangerous' function is one which results in a privilege escalation.
For example, if one were to read the channel variable SHELL(rm -rf /)
Bad Things(TM) could happen; even if the external source has only read
permissions.
Execution from external sources may be enabled by setting
'live_dangerously' to 'yes' in the [options] section of asterisk.conf.
Although doing so is not recommended.
Also, the ABI was changed to something more reasonable, since Asterisk
12 does not yet have a public release.
(closes issue ASTERISK-22905)
Review: http://reviewboard.digium.internal/r/432/
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This change adds an event for when an originated call is redirected to
another target. This event contains the original channel and the newly
created channel. If a stasis subscription exists on the original originated
channel for a stasis application then a new subscription will also be
created on the stasis application to the redirected channel. This allows
the application to follow the call path completely.
(closes issue ASTERISK-22719)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/3054/
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There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER
wouldn't be set on channels involved with blind and attended transfers.
This would happen with features that were initialized by channel driver
specific mechanisms in multiparty calls. This patch resolves those cases
while attempted to keep the behavior for setting those variables as
consistent as possible.
(closes issue AFS-24)
Review: https://reviewboard.asterisk.org/r/3040/
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The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
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Added the ability to have rules that are checked when adding and/or removing
channels to/from a bridge. In this case, if a channel is currently recording
and someone attempts to add it to a bridge an "is recording" rule is checked,
fails, and a 409 conflict is returned.
Also command functions now return an integer value that can be descriptive of
what kind of problems, if any, occurred before or during execution.
(closes issue ASTERISK-22624)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2947/
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In some cases messages need to be sent to a direct URI (sip:<ip address>). This
patch adds in that support by using a default outbound endpoint. When sending
messages, if no endpoint can be found then the default one is used.
To facilitate this a new default_outbound_endpoint option was added to the
globals section for pjsip.conf.
Review: https://reviewboard.asterisk.org/r/2944/
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* The voicemail registration/unregistration functions now take a struct of
callbacks instead of a lengthy parameter list of callbacks.
* The voicemail registration/unregistration functions now prevent a
competing module from interfering with an already registered callback
supplying module.
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This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
* RTP information, including source/destination media addresses, whether or
not the media is secure, held, and other properties.
* RTCP information. This includes sets of parseable information, as well as
individual statistic attriutes.
* PJSIP information. This includes URIs, local/remote signalling addresses,
whether or not the signalling is secure, and other properties.
* The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
function to obtain more detailed endpoint information.
Review: https://reviewboard.asterisk.org/r/3038/
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* Make ast_sorcery_observer_remove() accept a const callbacks struct.
* Make ast_sorcery_observer_remove() tolerant of the sorcery parameter
being NULL. Now it can be called within a module unload routine if the
sorcery initialization fails.
* Fix ast_sorcery_observer_add() to fail if the container link fails.
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This adds channel locks around calls to create channel snapshots as well
as other functions which operate on a channel and then end up
creating a channel snapshot. Functions that expect the channel to be
locked prior to being called have had their documentation updated to
indicate such.
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Transport type determination for security events has been simplified to use
the type present on the message itself instead of searching through configured
transports to find the transport used.
The actual WebSocket transport has also been simplified. It now leverages the
existing PJSIP transport manager for finding the active WebSocket transport
for outgoing messages. This removes the need for res_pjsip_transport_websocket
to store a mapping itself.
(closes issue ASTERISK-22897)
Reported by: Max E. Reyes Vera J.
Review: https://reviewboard.asterisk.org/r/3036/
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The patch allows ARI to parse request parameters from an incoming JSON
request body, instead of requiring the request to come in as query
parameters (which is just weird for POST and DELETE) or form
parameters (which is okay, but a bit asymmetric given that all of our
responses are JSON).
For any operation that does _not_ have a parameter defined of type
body (i.e. "paramType": "body" in the API declaration), if a request
provides a request body with a Content type of "application/json", the
provided JSON document is parsed and searched for parameters.
The expected fields in the provided JSON document should match the
query parameters defined for the operation. If the parameter has
'allowMultiple' set, then the field in the JSON document may
optionally be an array of values.
(closes issue ASTERISK-22685)
Review: https://reviewboard.asterisk.org/r/2994/
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Created a data model and implemented functionality for an ARI device state
resource. The following operations have been added that allow a user to
manipulate an ARI controlled device:
Create/Change the state of an ARI controlled device
PUT /deviceStates/{deviceName}&{deviceState}
Retrieve all ARI controlled devices
GET /deviceStates
Retrieve the current state of a device
GET /deviceStates/{deviceName}
Destroy a device-state controlled by ARI
DELETE /deviceStates/{deviceName}
The ARI controlled device must begin with 'Stasis:'. An example controlled
device name would be Stasis:Example. A 'DeviceStateChanged' event has also
been added so that an application can subscribe and receive device change
events. Any device state, ARI controlled or not, can be subscribed to.
While adding the event, the underlying subscription control mechanism was
refactored so that all current and future resource subscriptions would be
the same. Each event resource must now register itself in order to be able
to properly handle [un]subscribes.
(issue ASTERISK-22838)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3025/
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Created the following AMI commands and corresponding events for res_pjsip:
PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few
select attributes on each.
Events:
EndpointList - for each endpoint a few attributes.
EndpointlistComplete - after all endpoints have been listed.
PJSIPShowEndpoint - Provides a detail list of attributes for a specified
endpoint.
Events:
EndpointDetail - attributes on an endpoint.
AorDetail - raised for each AOR on an endpoint.
AuthDetail - raised for each associated inbound and outbound auth
TransportDetail - transport attributes.
IdentifyDetail - attributes for the identify object associated with
the endpoint.
EndpointDetailComplete - last event raised after all detail events.
PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound
registrations.
Events:
InboundRegistrationDetail - inbound registration attributes for each
registration.
InboundRegistrationDetailComplete - raised after all detail records have
been listed.
PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound
registrations.
Events:
OutboundRegistrationDetail - outbound registration attributes for each
registration.
OutboundRegistrationDetailComplete - raised after all detail records
have been listed.
PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions
and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound
subscriptions and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
(issue ASTERISK-22609)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2959/
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This change prevents channels used as implementation details from
leaking out to ARI. It does this by preventing creation of JSON blobs
of channel snapshots created from those channels and sanitizing JSON
blobs of bridge snapshots as they are created. This introduces a
framework for excluding information from output targeted at Stasis
applications on a consumer-by-consumer basis using channel sanitization
callbacks which could be extended to bridges or endpoints if necessary.
This prevents unhelpful error messages from being generated by
ast_json_pack.
This also corrects a bug where BridgeCreated events would not be
created.
(closes issue ASTERISK-22744)
Review: https://reviewboard.asterisk.org/r/2987/
Reported by: David M. Lee
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SIP transaction group lock support has been backported into our pjproject. Since the code
now internally uses a group lock the code is now changed to unlock it if present. Note
that the act of finding the transaction is what actually returns it locked.
For further information about group locks check out the wiki page at:
http://trac.pjsip.org/repos/wiki/Group_Lock
(issue ASTERISK-22818)
Reported by: Matt Jordan
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This patch explicitly defines AST_AF_* enum constants to their sys/socket.h
defined equivalents. It is certainly unclear why these constants actually have
to exist, given that netsock2.h includes sys/socket.h; however, since the code
base is already liberally sprinkled with the usage of AST_AF_* (as well as with
direct calls to AF_*), this will at least keep the semantics consistent between
their usage across systems.
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Made the vector macro API be more like linked lists.
1) Added a name parameter to ast_vector() to name the vector struct.
2) Made the API take a pointer to the vector struct instead of the struct
itself.
3) Added an element cleanup macro/function parameter when removing an
element from the vector for ast_vector_remove_cmp_unordered() and
ast_vector_remove_elem_unordered().
4) Added ast_vector_get_addr() in case the vector element is not a simple
pointer.
* Converted an inline vector usage in stasis_message_router to use the
vector API. It needed the API improvements so it could be converted.
* Fixed topic reference leak in router_dtor() when the
stasis_message_router is destroyed.
* Fixed deadlock potential in stasis_forward_all() and
stasis_forward_cancel(). Locking two topics at the same time requires
deadlock avoidance.
* Made internal_stasis_subscribe() tolerant of a NULL topic.
* Made stasis_message_router_add(),
stasis_message_router_add_cache_update(), stasis_message_router_remove(),
and stasis_message_router_remove_cache_update() tolerant of a NULL
message_type.
* Promoted a LOG_DEBUG message to LOG_ERROR as intended in
dispatch_message().
Review: https://reviewboard.asterisk.org/r/2903/
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* Typedefed and added doxegen for the voicemail callback functions.
* Simplified the prototypes for ast_install_vm_functions() and
ast_install_vm_test_functions() to use the new function typedefs.
* Simplified the voicemail callback function pointer variable declarations
to use the new function typedefs.
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Adds the following AMI events, closely following their CLI counterparts:
BridgeDestroy
BridgeKick
BridgeTechnologyList
BridgeTechnologySuspend
BridgeTechnologyUnsuspend
BridgeDestroy kicks an entire bridge, where BridgeKick kicks just one
channel off the bridge. When kicking a channel, specifying the bridge
also (optional) insures it is not removed from the wrong bridge. The
BridgeTechnology events allow viewing and changing suspension status,
which affects only subsequent not active bridging.
(closes ASTERISK-22356)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/2973/
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The ring operation sends ringing to the specified channel it is invoked on.
The dtmf operation can be used to send DTMF digits to the specified channel
of a specific length with a wait time in between. Finally hangup reasons
allow you to specify why a channel is being hung up (busy, congestion).
Early media behavior has also been tweaked slightly. When playing media to a channel
it will no longer automatically answer. If it has not been answered a progress indication
is sent instead.
(closes issue ASTERISK-22701)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2916/
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Merged revisions 402358 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402359 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If a Stasis application is specified an implicit subscription is done on the originated
channel. This was previously done with the channel lock held which is dangerous as the
underlying code locks the container and iterates items. This change releases the lock
on the originated channel before subscribing occurs.
(closes issue ASTERISK-22768)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2979/
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Merged revisions 402346 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
