Commit graph

2311 commits

Author SHA1 Message Date
Richard Mudgett
3357c494cb sorcery, bucket: Change observer remove calls to take const callbacks struct.
* 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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Merged revisions 403324 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-12-03 17:35:54 +00:00
Mark Michelson
8e8b329e14 Add channel locking for channel snapshot creation.
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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2013-12-03 17:07:29 +00:00
Joshua Colp
177e7861a2 res_pjsip_session: Apply fromuser and fromdomain to all requests as documented.
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Merged revisions 403271 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-12-01 21:13:20 +00:00
Joshua Colp
88c840db50 res_pjsip_t38: Add the framehook to the channel only on first INVITE.
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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Merged revisions 403258 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-12-01 20:04:55 +00:00
Joshua Colp
0620cc0c00 res_pjsip_transport_websocket: Fix security events and simplify implementation.
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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Merged revisions 403256 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-12-01 19:58:08 +00:00
Joshua Colp
e93fbf41e6 res_ari: Add Recording events to the validator.
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Merged revisions 403240 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-30 14:12:50 +00:00
Joshua Colp
56290895aa res_pjsip_sdp_rtp: Don't produce an invalid media stream with no formats.
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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Merged revisions 403223 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-28 02:12:45 +00:00
Joshua Colp
b315b16c90 res_pjsip_header_funcs: Don't add headers to re-INVITEs.
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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Merged revisions 403221 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-28 01:56:52 +00:00
Joshua Colp
6019353ad6 res_stasis_playback: Add 'number', 'digits', and 'characters' URI scheme 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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Merged revisions 403209 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-28 00:54:37 +00:00
Joshua Colp
a64cd7c6bb res_pjsip_session: Add configurable behavior for redirects.
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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Merged revisions 403207 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-28 00:38:36 +00:00
Joshua Colp
9d3b590ad8 res_pjsip: Fix crash when reloading certain configurations.
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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Merged revisions 403179 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-27 16:12:56 +00:00
David M. Lee
fccb427c88 ari:Add application/json parameter support
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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2013-11-27 15:48:39 +00:00
Joshua Colp
fd33969240 res_pjsip: Update handling of some options to work with new option names.
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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2013-11-27 15:31:43 +00:00
Kevin Harwell
ed48377994 ARI: Implement device state API
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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2013-11-23 17:48:28 +00:00
Kevin Harwell
05cbf8df9b res_pjsip: AMI commands and events.
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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2013-11-23 17:26:57 +00:00
Joshua Colp
14a7452934 ari: Add events for playback and recording.
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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Merged revisions 403119 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-23 12:52:54 +00:00
Joshua Colp
eda7126862 ari: Add Snoop operation for spying/whispering on channels.
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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2013-11-23 12:40:46 +00:00
Kevin Harwell
76a2b855e1 res_pjsip: convert configuration settings names to snake case some more
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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2013-11-22 22:37:30 +00:00
Kinsey Moore
d9015a5356 ARI: Don't leak implementation details
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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2013-11-22 20:10:46 +00:00
Kevin Harwell
1c45a32ee8 res_pjsip: convert configuration settings names to snake case
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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2013-11-22 17:27:55 +00:00
Matthew Jordan
92af2b2e26 res_pjsip_session: Fix memory leak of direct media format capabilities
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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2013-11-21 19:22:18 +00:00
Matthew Jordan
c3575e338e res_pjsip_sdp_rtp: Fix use of uninitialized value in PJSIP
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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2013-11-21 17:53:39 +00:00
Matthew Jordan
08686e62c5 res_pjsip_mwi: Fix memory leak of MWI subscriptions container
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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2013-11-21 17:53:22 +00:00
David M. Lee
f0ccc59a22 stasis: Fixed scoping problem with bridge tracking.
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2013-11-21 15:57:40 +00:00
David M. Lee
d1ad4a95f8 ari: Add silence generator controls
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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Merged revisions 402926 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-21 15:56:34 +00:00
Joshua Colp
71612fb007 res_pjsip_caller_id: Don't overwrite user portion of the From header when 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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2013-11-19 23:17:57 +00:00
Joshua Colp
1b14a78d14 res_pjsip: Add support for building against pjproject with SIP transaction 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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2013-11-16 13:51:04 +00:00
Joshua Colp
67b650543c res_ari_channels: Add the ability to stop locally generated ringing on a 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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2013-11-13 23:11:32 +00:00
Kevin Harwell
f6593b4156 ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404
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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2013-11-12 23:17:45 +00:00
Kevin Harwell
4be01b4598 ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404
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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2013-11-12 22:17:28 +00:00
Kevin Harwell
12a0edac69 pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer dereferences
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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2013-11-12 16:49:17 +00:00
David M. Lee
5701c51bb6 Fixed a typ.
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2013-11-12 15:27:00 +00:00
Richard Mudgett
7323809607 res_stasis.c: Fix locking issues with the app_bridge_moh container.
* 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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2013-11-08 20:37:08 +00:00
Mark Michelson
518f091a1a Clarify an ambiguous error message.
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2013-11-08 19:22:53 +00:00
David M. Lee
4c128198c8 res_pjsip: Print a helpful error message if sorcery registration fails
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2013-11-08 18:53:14 +00:00
David M. Lee
b83a3965b8 Changes from make ari-stubs after r402560
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2013-11-08 18:52:19 +00:00
Kevin Harwell
4f1bdeed1c ARI playback: Rename ARI Playback to Playbacks
Before playback was the only non plural resource.  It has been renamed to
playbacks for consistency.

(closes issue ASTERISK-22737)
Reported by: Paul Belanger
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2013-11-08 17:59:16 +00:00
David M. Lee
97a8debd90 ari: Add application/x-www-form-urlencoded parameter support
ARI POST calls only accept parameters via the URL's query string.
While this works, it's atypical for HTTP API's in general, and
specifically frowned upon with RESTful API's.

This patch adds parsing for application/x-www-form-urlencoded request
bodies if they are sent in with the request. Any variables parsed this
way are prepended to the variable list supplied by the query string.

(closes issue ASTERISK-22743)
Review: https://reviewboard.asterisk.org/r/2986/
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2013-11-08 17:29:53 +00:00
Jonathan Rose
3c645e8520 PJSIP: Improve error handling in digest authenticator
Previously, regardless of whether failure to authenticate was due to
lacking any authentication or actually failing authentication, the
Digest Authenticator would simply return that a challenge was still
needed. It will continue to do that when no authentication information
is in the received SIP digest, but when authentication information
is present and does not pass authentication, that will be treated as
an authentication error. This is to ensure that PJSIP will issue
security events indicated failed auths.
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2013-11-07 23:42:31 +00:00
David M. Lee
7d0d1a1efb ari: User better nicknames for ARI operations
While working on building client libraries from the Swagger API, I
noticed a problem with the nicknames.

    channel.deleteChannel()
    channel.answerChannel()
    channel.muteChannel()

Etc. We put the object name in the nickname (since we were generating C
code), but it makes OO generators redundant.

This patch makes the nicknames more OO friendly. This resulted in a lot
of name changing within the res_ari_*.so modules, but not much else.

There were a couple of other fixed I made in the process.

 * When reversible operations (POST /hold, POST /unhold) were made more
   RESTful (POST /hold, DELETE /unhold), the path for the second operation
   was left in the API declaration. This worked, but really the two
   operations should have been on the same API.
 * The POST /unmute operation had still not been REST-ified.

Review: https://reviewboard.asterisk.org/r/2940/
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2013-11-07 21:10:31 +00:00
Joshua Colp
7678fd040e res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media.
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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2013-11-01 14:38:21 +00:00
Kinsey Moore
98dea21bc1 chan_sip: Fix RTCP port for SRFLX ICE candidates
This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.

(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
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2013-11-01 12:40:40 +00:00
Joshua Colp
4053f36a71 res_ari_channels: Fix a deadlock when originating multiple channels close to eachother.
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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2013-11-01 12:33:09 +00:00
Joshua Colp
d17a780333 res_stasis: Ensure the channel is always departed from the bridge when it leaves.
This change adds a command to the command queue to explicitly depart the channel
from the bridge when it is told it has left. If the channel has already been departed
or has entered a different bridge this command will become a no-op.

(closes issue ASTERISK-22703)
Reported by: John Bigelow

(closes issue ASTERISK-22634)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2965/
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2013-11-01 12:13:09 +00:00
David M. Lee
069da1e75a stasis: add functions embarrassingly missing from r400522
I neglected to implement two of the endpoint subscription functions when
I did the work. Normally, you'll only hit that when you unsubscribe from
a specific endpoint.
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2013-10-31 14:45:03 +00:00
Kevin Harwell
4746c068dc pjsip_messaging: Added debug for in dialog messaging
(issue ASTERISK-22777)
Reported by: Matt Jordan
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2013-10-30 17:54:26 +00:00
Kinsey Moore
aa7f9e55f2 ARI: Remove channels/{channelId}/dial
This removes the /ari/channels/{channelId}/dial URI since it is
redundant, overly complex, is likely to become more externally complex
over time, and is too high-level compared with other ARI operations.
See the following for further information:
http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html

(closes issue ASTERISK-22784)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2968/
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2013-10-29 12:51:57 +00:00
Joshua Colp
fd98037fe2 res_ari_playback: Add missing 404 error response for GET and DELETE.
(closes issue ASTERISK-22722)
Reported by: Richard Mudgett
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2013-10-29 11:15:59 +00:00
Richard Mudgett
8e4084b586 res_stasis.c: Made use the ao2_container callback templates.
* Made res_stasis.c use the OBJ_SEARCH_XXX defines.
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2013-10-26 00:36:31 +00:00
Richard Mudgett
7eea4ab872 You'd think that new files would be free of whitespace issues. But you would be wrong.
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2013-10-25 22:03:04 +00:00