Age | Commit message (Collapse) | Author |
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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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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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When switching to using a vector for authentication, I initialized
the vector for the artificial endpoint to be of size 1. However, this
does not result in AST_VECTOR_SIZE() returning 1 since there isn't
actually anything in the vector.
Rather than trifle with the vector by putting unnecessary elements in,
I simply changed the callback in res_pjsip_authenticator_digest.c to
explicitly report that the artificial endpoint requires authentication.
Thanks to Joshua Colp for pointing this out.
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res_sorcery_astdb.c: Fix get multiple records by regex.
* Fix sorcery_astdb_retrieve_regex() pattern matching. Let the regexec()
function match the stored key values instead of having astdb prefilter
them. Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.
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* Fix sorcery_astdb_retrieve_regex() pattern matching. Let the regexec()
function match the stored key values instead of having astdb prefilter
them. Previoiusly you could only use a simple regex pattern when the
pattern began with '^'.
* Fix off nominal memory leak in sorcery_astdb_retrieve_regex().
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Due to the way pjproject internally works it was possible for the
NAT module to not be invoked on messages with-in a session dialog.
This means that the various parts of the message would not get rewritten
with the source IP address and port.
This change uses a session supplement to add the NAT module
to the dialog on the first incoming or outgoing INVITE.
(closes issue ASTERISK-22941)
Reported by: Leif Madsen
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Since Asterisk has a vector API now, places where arrays are manually
resized don't really make sense any more. Since the auth work in PJSIP
was freshly-written, it was easy to reform it to use a vector.
Review: https://reviewboard.asterisk.org/r/3044
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Prior to this patch, res_fax_spandsp was conservative with how it initialized
the spandsp T.38 context. It would only initialize it if the driver thought
the current state was a T.38 fax. While this works fine in nominal situations,
in certain off nominal situations, res_fax_spandsp can believe that a T.38
fax will not occur when in fact one has started. In particular, this was
discovered when res_fax would fall back to audio after timing out on a T.38
upgrade. The SIP channel driver would continue to retry the re-INVITE and -
if the remote end responded after res_fax timed out with a 200 OK - a T.38
frame would be delivered to the res_fax stack when it no longer expected it.
As it turns out, there does not appear to be any downside to always
initializing the T.38 context, other than the actual memory allocation.
Since that avoids this off nominal situation (and others which are equally
likely hard to predict), this is the safest way to avoid this problem.
Much thanks to Torrey as well for providing a scenario that reproduces this
issue.
(closes issue ASTERISK-21242)
Reported by: Ashley Winters
Tested by: Torrey Searle
patches:
always-init-t38.patch uploaded by awinters (License 6477)
A_PARTY.xml uploaded by tsearle (License 5334)
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If the CDR unregistration fails due to an inflight CDR, the
res_config_sqlite module needs to bail on unloading itself. Otherwise,
the config could be unloaded (including the CDR table name) while the
CDR engine posts a CDR to the still registered backend, resulting in
a crash.
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The code for getting channel variables from ARI assumed that you needed
to lock the channel in order to properly execute functions and read
channel variables. Apparently, this is not the case, since any dialplan
function that puts the channel into autoservice deadlocks when
attempting to remove the channel from autoservice.
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Used a static wrapper around the offending function to alleviate the issue.
Reported by: rmudgett
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This crept up during gateway testing where the gateway would receive
the request to negotiate and assume it came from the remote side, causing
the gateway state machine to go a little, to a use a technical term,
"wonky".
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Passing a non-zero value causes PJLIB to use the given input as the
hash value. Passing zero causes the parameter to become an output parameter
that receives the hash value that was computed based on the given key.
This change essentially makes ast_sip_dict_get() properly retrieve the
desired value.
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Newer versions of PJSIP have changed to using a flag for the
PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a
configure check to detect the presence of the flag and use it if found.
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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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The check for determining whether the T.38 framehook should be added to
the channel or not has now been changed to guarantee adding only occurs
on the first incoming or outgoing INVITE.
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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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Depending on configuration it was possible for a media stream to be
created without any media formats. The produced SDP would fail internal
validation and cause a crash.
The code will now no longer add media streams with no formats to the SDP,
allowing it to pass validation and work.
(closes issue ASTERISK-22858)
Reported by: Anthony Messina
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When sending a re-INVITE to an endpoint it was possible for received
headers to be added as well (since they are stored for retrieval using
the PJSIP_HEADER dialplan function). This caused a broken (and
potentially large) SIP INVITE to be produced and sent.
This changes the module so it will no longer add headers to
re-INVITEs.
(closes issue ASTERISK-22882)
Reported by: David M. Lee
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implementations.
This change adds new URI scheme implementations for playing numbers, digits,
and characters. This is done as part of the normal playback mechanism and can
be used with queueing to create a combined sentence.
Review: https://reviewboard.asterisk.org/r/3028/
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The action taken when a redirect occurs is now configurable on a
per-endpoint basis. The redirect can either be treated as a redirect
to a local extension, to a URI that is dialed through the Asterisk
core, or to a URI that is dialed within PJSIP itself.
(closes issue ASTERISK-21710)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2963/
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Certain options available that specify a SIP URI perform validation
on the provided URI using the PJSIP URI parser. This operation
requires that the thread executing it be registered with the PJLIB
library. During reloads this was done on a thread which was NOT
registered with it.
This fixes the problem by creating a task which reloads the
configuration on a PJSIP thread.
(closes issue ASTERISK-22923)
Reported by: Anthony Messina
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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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Some options (such as call_group and pickup_group) share the same configuration
handler and decide what logic to use based on the name of the option. These
handlers were not updated to check for the new option names and were treating
the options as invalid.
This change simply updates the handlers with the proper names of the options.
(closes issue ASTERISK-22922)
Reported by: Anthony Messina
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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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While there were events defined for playback and recording
these were not actually sent. This change implements the
to_json handlers which produces them.
(closes issue ASTERISK-22710)
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/3026/
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The Snoop operation can be invoked on a channel to spy or
whisper on it. It returns a channel that any channel operations
can then be invoked on (such as record to do monitoring).
(closes issue ASTERISK-22780)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3003/
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Updated the alembic script for pjsip. Also, the dtls config parsing stuff was
expecting strings with no underscores, so removed the underscores from the
option name before passing it to the parser.
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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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Renamed, where appropriate, the configuration options for chan/res_pjsip to use
snake case (compound words separated by an underscore). For example, faxdetect
will become fax_detect, recordofffeature will become record_off_feature, etc...
Review: https://reviewboard.asterisk.org/r/3002/
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The direct media format capabilities are always allocated in
ast_sip_session_alloc and were not freed in the session destructor. Whoops.
(This being the third whoops caught by Scott and Nitesh's valgrind work for
the Asterisk Test Suite. Nifty!)
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In PJMEDIA, pjmedia_sdp_rtpmap_to_attr will attempt to use the string
rtpmap.param regardless of its length value. Simply setting the length to 0
does not prevent the garbage on the stack in rtpmap.param.ptr from being
formatted in a sprintf call. This patch initializes the string to NULL so that
at the very least, something is provided to the function that is predictable.
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This patch fixes a reference counting memory leak on the ao2_container
created as part of create_mwi_subscriptions. When we create the container
in this routine, the intent is to hand lifetime ownership over to the global
container unsolicited_mwi. When ao2_global_obj_replace_unref is called, the
reference count on mwi_subscriptions (the container) will be bumped by 1;
however, the function does not decrement the reference count on
mwi_subscriptions when this occurs. This will prevent the container from being
fully disposed of when Asterisk exits (or on any subsequent call to this
operation, such as during a reload).
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This patch adds the ability to start a silence generator on a channel
via ARI. This generator will play silence on the channel (avoiding audio
timeouts on the peer) until it is stopped, or some other media operation
is started (like playing media, starting music on hold, etc.).
(closes issue ASTERISK-22514)
Review: https://reviewboard.asterisk.org/r/3019/
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fromuser is set.
The fromuser option is used to explicitly set the user within the From header. The
res_pjsip_caller_id module did not take this setting into account when determining
if the From header could be modified or not.
(closes issue ASTERISK-22866)
Reported by: Anthony Messina
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group lock support.
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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channel.
Using the 'ring' operation it is possible to start locally generated ringback if
the channel is answered. This change adds the ability to stop it by using DELETE.
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Was returning a 404 on a valid technology with an empty list of endpoints.
Now checking against the channel tech to make sure the tech itself is valid
and not just an empty list of endpoints.
(issue ASTERISK-22803)
Reported by: David M. Lee
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Implementation listing endpoints by technology returned an empty array if no
matching endpoints were found. Fixed so a "404 Not Found" will be returned
instead.
(closes issue ASTERISK-22803)
Reported by: David M. Lee
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Both res_pjsip_messaging and res_pjsip_header_funcs were causing asterisk to
crash because they were trying to dereference a NULL pointer.
In the case of res_pjsip_messaging it was attempting to "print" a contact
header that did not exist. In fact contact headers should not be part of
a SIP MESSAGE, so the offending code was simply removed.
In the case of res_pjsip_header_funcs a null private channel tech was being
passed to the function and then later dereferenced. Added null checks (and
error logging) to the read/write function handlers to guard against crashing.
(closes issue ASTERISK-22821)
Reported by: Anthony Messina
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* Fix unlinking from the app_bridges_moh container in remove_bridge_moh()
without a lock under normal circumstances.
* Made check ast_bridge_set_after_callback() return value in
bridge_moh_create() to handle failure.
* Fixed SCOPED_AO2LOCK() locking over too much scope in
stasis_app_bridge_moh_channel() and stasis_app_bridge_moh_stop().
* Fixed unusual usage of ao2_unlink_flag() in control_unlink().
* Fixed orphaned bridge from off nominal path in
stasis_app_bridge_create().
* Fixed strange construct in stasis_app_unsubscribe(). From a bad merge?
* Made load_module() cleanup on failure.
Review: https://reviewboard.asterisk.org/r/2962/
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