Age | Commit message (Collapse) | Author |
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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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The configure check did not use the provided paths for pjproject
if provided when looking for transaction group lock support.
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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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vm-msgsaved
In the last release of sounds, 1.4.25 we added a vm-msgforwarded prompt for various core languages. Now we use that prompt.
(issue ASTERISK-21413)
(closes issue ASTERISK-21413)
Reported by: netwrkr
Tested by: newtonr
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This fixes the unit tests that were broken by r403069 and several
functions requiring a new parameter for sanitization of JSON messages
generated from object snapshots.
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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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When translating from one format to another it is possible
to inform the translation function that the source frame should
be freed. This was previously done immediately but shortly
afterwards the frame that was freed was accessed and used again.
This change moves code around a bit so that the frame is now
freed after it has been completely used.
(closes issue ASTERISK-22788)
Reported by: Corey Farrell
Patches:
translate-access-after-free-11up.patch uploaded by coreyfarrell (license 5909)
translate-access-after-free-1.8.patch uploaded by coreyfarrell (license 5909)
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* Made PickupChan() search by channel uniqueids if the search could not
find a channel by name.
* Ensured PickupChan() never considers the picking channel for pickup.
* Made PickupChan() option p use a common search by name routine. The
original search was erroneously case sensitive.
(issue AFS-42)
Review: https://reviewboard.asterisk.org/r/3017/
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By the time the directory application exits, a channel variable
DIRECTORY_RESULT will be set for the channel that invoked it which
can be used to determine the reason for exit. The changes log and
the app_directory documentation contain specific details about
each of the possible values for DIRECTORY_RESULT.
Review: https://reviewboard.asterisk.org/r/3016/
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For the new deviceState resource, we need to properly generate
device_state.[ch] files.
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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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Review: https://reviewboard.asterisk.org/r/3008/
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This fixes a crash when CELGenUserEvent is called from the dialplan
while CEL is disabled. Currently, CEL does not create its topics and
forwards if it is not enabled and external entities may depend on
these topics blindly since they should always be available. This patch
breaks up route creation and topic/forward creation such that the CEL
topics and forwards will always exist while the router and its
associated routes will be torn down and recreated as necessary.
(closes issue ASTERISK-22799)
Review: https://reviewboard.asterisk.org/r/3010/
Reported by: Matt Jordan
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* Made Pickup() and PickupChan() tollerate empty pickup values. i.e., You
can now have Pickup(&&exten@context).
* Made PickupChan() use the standard option flag parsing code.
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Similar to how background works, if a say application is called with
this variable set to 'true', 'yes', 'on', etc. then using DTMF while
the say action is in progress will result in the channel jumping to
that extension in the dialplan.
Review: https://reviewboard.asterisk.org/r/3011/
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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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Two variables were being checked for NULLity immediately
after being declared NULL. I moved the NULL check until
after the variables are allocated.
This allows for the "channelvars" option in manager.conf
to work as intended again.
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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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This prevents NULL from being passed into an ast_json_pack call when no
extra information is passed to the application which prevents an error
message about NULL arguments from being generated.
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Asterisk will sometimes core dump during caller id read on analog
channels due to a negative return value from the read() in
my_get_callerid that slips through as a negative length argument to
callerid_feed() if the errno returned by DAHDI is ELAST. This change
ensures that the negative return is treated properly even when it is
ELAST.
(closes issue ASTERISK-22746)
Reported by: Michael Walton
Patches:
chan_dahdi_cid_crash_fix.r401410.patch uploaded by Michael Walton (License 6502)
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Adds a couple of test events for conference menu actions so that it's
easy to discern when those menu actions have been triggered.
(issue ASTERISK-22760)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2999/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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I'm doing some unrelated work in app_confbridge and finding
these "invalid pin" comments to be annoying. Get out!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In the current app_queue code from 1.8 up to trunk the upper and lower
penalties can be set to 0 but the value is interpreted to be disabled
instead of actually setting limits. This is especially evident if min
and max limits are set to 0 and members with penalties of 0 and 1 are
in the queue since the member with penalty 1 will still receive calls.
This patch adjusts the special disabled value to be INT_MAX instead of
0.
(closes issue ASTERISK-20862)
Review: https://reviewboard.asterisk.org/r/2995/
Reported by: Schmooze Com
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402648 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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For outbound register requests the tag on the From line was
updated every 20 seconds prior to a successful registration
and also once for each registration renewal. That behavior
can possibly cause the registration to be denied because of
the different tag, and is not aligned with the intention of
RFC 3261 8.1.3.5 "... request constitutes a new transaction
and SHOULD have the same value of the Call-ID, To, and From
of the previous request...". This updates chan_sip to have
a field to keep the local tag in the registration structure
and use that tag for registration requests where the callid
is also unchanged.
(closes issue ASTERISK-12117)
Reported by: Pawel Pierscionek
Review: https://reviewboard.asterisk.org/r/2988/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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