Commit Graph

2364 Commits

Author SHA1 Message Date
Jonathan Rose
21e22310c7 ARI: Music on Hold/Background Music for bridges
Adds ARI functions to be able to turn on/off music on hold in a
bridge. It actually functions more as a background music without
further actions on the bridge since if the rest of the channels
in the bridge aren't explicitly muted, they will still be able
to communicate.

(closes issue ASTERISK-21974)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2688/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23 00:26:19 +00:00
Kinsey Moore
7b032c1adb Add SayAlphaCase and similar functionality for AGI
This adds a new dialplan application, SayAlphaCase, that performs much
the same function as SayAlpha except that it takes additional options
which allow the user to specify whether the case of each letter should
be announced for uppercase, lowercase, or all letters. Similar
functionality has been added to the SAY ALPHA AGI command via an
optional parameter.

Original Patch by: Kevin Scott Adams
Reported by: Kevin Scott Adams
Review: https://reviewboard.asterisk.org/r/2725/
(closes issue ASTERISK-20782)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 22:33:48 +00:00
Kevin Harwell
aefebebd37 res_sip_dtmf_info: Support sending of 'raw' DTMF
Added the ability to handle 'raw' DTMF within the body of an INFO message.
Also made it so values 10-16 are mapped to valid DTMF values.

(closes issue ASTERISK-22144)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2776/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 22:09:16 +00:00
Kinsey Moore
6fa4e8e3ab Add missing configOption close tags
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 21:39:10 +00:00
Rusty Newton
9520df86fd Fix missing xml doc configOption 'type' for for both 'system' and 'global' configObjects
(issue ASTERISK-22344)
(closes issue ASTERISK-22344)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 21:21:25 +00:00
Richard Mudgett
477dea4661 Bridge API: Set a cause code on a channel when it is ejected from a bridge.
The cause code needs to be passed from the disconnecting channel to the
bridge peers if the disconnecting channel dissolves the bridge.

* Made the call to an app_agent_pool agent disconnect with the busy cause
code if the agent does not ack the call in time or hangs up before acking
the call.

(closes issue ASTERISK-22042)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2772/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 21:09:52 +00:00
Mark Michelson
8049bf94f7 Handle default body types for SIP event packages in res_pjsip_pubsub
Prior to this change, we would reject SUBSCRIBE requests that had no Accept
headers. Now event package handlers that handle the default type for the
event package indicate that they do so. Therefore, if we have a handler that
can handle the default type, we can allow SUBSCRIBEs for the handler's event
package that have no Accept headers.

(closes issue ASTERISK-22067)
reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/2774


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 17:42:37 +00:00
David M. Lee
5762c1b4ac ARI: Correct segfault with /variable calls are missing ?variable parameter.
Both /asterisk/variable and /channel/{channelId}/variable requires a
?variable parameter to be passed into the query. But we weren't checking
for the parameter being missing, which caused a segfault.

All calls now properly return 400 Bad Request errors when the parameter
is missing. The Swagger api-docs were updated accordingly.

(closes issue ASTERISK-22273)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21 16:23:59 +00:00
David M. Lee
f5cca5e41e res_stasis: remove call to missing function control_continue.
In the shuffling around of res_stasis, control_continue was renamed to
stasis_app_control_continue, but the call in res_stasis wasn't updated.
In looking into it, it turns out it wasn't really the right thing to do
in res_stasis anyways.

This patch changes the handling of received a AST_CONTROL_HANGUP frame
to be the same as receiving a NULL frame, and removed the declaration of
control_continue(), since it doesn't exist any more.

(closes issue ASTERISK-22292)
Reported by: Denis Smirnov


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21 16:00:10 +00:00
Richard Mudgett
d213dfa30f Fix several interrelated issues dealing with the holding bridge technology.
* Added an option flags parameter to interval hooks.  Interval hooks now
can specify if the callback will affect the media path or not.

* Added an option flags parameter to the bridge action custom callback.
The action callback now can specify if the callback will affect the media
path or not.

* Made the holding bridge technology reexamine the participant idle mode
option whenever the entertainment is restarted.

* Fixed app_agent_pool waiting agents needlessly starting and stopping MOH
every second by specifying the heartbeat interval hook as not affecting
the media path.

* Fixed app_agent_pool agent alert from restarting the MOH after the alert
beep.  The agent entertainment is now changed from MOH to silence after
the alert beep.

* Fixed holding bridge technology to defer starting the entertainment.  It
was previously a mixture of immediate and deferred.

* Fixed holding bridge technology to immediately stop the entertainment.
It was previously a mixture of immediate and deferred.  If the channel
left the bridging system, any deferred stopping was discarded before
taking effect.

* Miscellaneous holding bridge technology rework coding improvements.

Review: https://reviewboard.asterisk.org/r/2761/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21 15:51:19 +00:00
Kinsey Moore
d12350ccc3 Allow channels in app_stasis to hangup properly
This detects hangups that occur while bridged to allow channels to exit
app_stasis even if the hangup frame was absorbed by the bridge the
channel was in.

Reported by: David Lee
(closes issue ASTERISK-22297)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-21 14:08:23 +00:00
Mark Michelson
5caa938be2 Localize and rename ACL configuration.
This is more-or-less a reversion of previous ACL behavior so that
it is more self-contained. ACL sections are now only parsed if res_pjsip_acl.so
is loaded. Moreover, the configuration section is now "type=acl" instead of
"type=security".

The original reason for having ACLs configured in a "type=security" section
was to lump ACLs and other security-related items into the same section. The
problem is that ACLs really should be in their own sections and there are
no other security-related options implemented anyways.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397193 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 21:01:59 +00:00
Mark Michelson
86741bdf46 Clarify documentation for the "identify_by" option for SIP endpoints.
This also removes documentation for the options that no longer exist.

(closes issue ASTERISK-22306)
reported by Rusty Newton


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 15:39:38 +00:00
Mark Michelson
ed19b8ee76 Add debug message to res_pjsip_endpoint_identifier_ip to indicate when an endpoint is successfully retrieved.
(closes issue ASTERISK-22101)
reported by Rusty Newton



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 15:32:20 +00:00
Mark Michelson
8931502f7a Add warning messages for registration failure paths.
(closes issue ASTERISK-22089)
reported by Rusty Newton
patches:
	patch1.txt uploaded by John Bigelow (License #5091)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 15:27:48 +00:00
Mark Michelson
14ba1751f6 Add note to transport configuration that a restart is required to change transports.
(closes issue ASTERISK-22094)
reported by Rusty Newton



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 14:43:56 +00:00
Joshua Colp
17f332169c Remove assumption in res_pjsip_dtmf_info that all INFO messages will contain a body.
(closes issue ASTERISK-22320)
Reported by: Matt Jordan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-20 11:33:43 +00:00
Kinsey Moore
dbe1520f35 Disable build of res_corosync until it is back in a compiling state
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396908 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-18 02:55:54 +00:00
Rusty Newton
168c679e6c xml doc changes for 'aor' config object and a few of its options
Added or modified text in the xml doc for the 'aor' config object to address a few issues:
* help for the 'mailboxes' option didn't make it clear how the "list" should be formatted.
* AoR object's involvement in inbound registration wasn't mentioned.
* help for the 'contact' option didn't describe how to specify multiple contacts.
* help for the 'max_contacts' option didn't tell whether it limited the amount of contacts defined through static configuration.

(issue ASTERISK-22118)
(closes issue ASTERISK-22118)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 18:13:11 +00:00
Rusty Newton
6b1f91b6fc 'domain_alias' config object XML help doesn't make it clear that the name used for the object is the domain alias
(issue ASTERISK-22114)
(closes issue ASTERISK-22114)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:47:34 +00:00
Rusty Newton
fdfe1ea82e xml doc changes for clarity - 'auth' config object and auth's 'auth_type' config option
(issue ASTERISK-22108)
(closes issue ASTERISK-22108)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:40:34 +00:00
Rusty Newton
115c078e9d xml doc change for transport config object - remove non-applicable warning and add text regarding Asterisk restart
(closes issue ASTERISK-22105)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 17:27:40 +00:00
Kinsey Moore
56aea1c030 Allow res_parking to be unloadable
This change protects accesses of res_parking such that it can unload
safely once transient uses of its registered functions are complete.
The parking API has been restructured such that its consumers do not
have access to the vtable exposed by the parking provider, but instead
route through stubs to prevent consumers from holding on to function
pointers.

This adds calls to all the parking unload functions and moves
application loading and unloading into functions in
parking_applications.c similar to the rest of the parts of res_parking.

Review: https://reviewboard.asterisk.org/r/2763/
(closes issue ASTERISK-22142)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 15:01:54 +00:00
Kinsey Moore
59753b1ea1 Strip down the old event system
This removes unused code, event types, IE pltypes, and event IE types
where possible and makes several functions private that were once
public. This includes a renumbering of the remaining event and IE types
which breaks binary compatibility with previous versions. The last
remaining consumers of the old event system (or parts thereof) are
main/security_events.c, res/res_security_log.c, tests/test_cel.c,
tests/test_event.c, main/cel.c, and the CEL backends.

Review: https://reviewboard.asterisk.org/r/2703/
(closes issue ASTERISK-22139)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-17 14:39:27 +00:00
Richard Mudgett
e47d3db365 Doxygen comment tweaks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-16 17:33:21 +00:00
Richard Mudgett
8b7742202f Minor parking cleanup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15 21:52:01 +00:00
Richard Mudgett
6b062d9afd Parking: Eliminate local channel name hack to get peer channel.
(closes issue ASTERISK-22034)
Reported by: Matt Jordan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15 20:09:10 +00:00
Kinsey Moore
e9ac63f9a9 Prevent automagic things from happening to Stasis application bridges
This prevents swap optimization, merges, and transfers involving Stasis
application bridges. It wouldn't be nice if the bridge you thought you
owned disappeared from under you.

Reported-by: Richard Mudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-15 12:05:41 +00:00
David M. Lee
987fdfb444 ARI: allow other operations to happen while bridged
This patch changes ARI bridging to allow other channel operations to
happen while the channel is bridged.

ARI channel operations are designed to queue up and execute
sequentially. This meant, though, that while a channel was bridged,
any other channel operations would queue up and execute only after the
channel left the bridge.

This patch changes ARI bridging so that channel commands can execute
while the channel is bridged. For most operations, things simply work
as expected. The one thing that ended up being a bit odd is recording.

The current recording implementation will fail when one attempts to
record a channel that's in a bridge. Note that the bridge itself may
be recording; it's recording a specific channel in the bridge that
fails. While this is an annoying limitation, channel recording is
still very useful for use cases such as voice mail, and bridge
recording makes up much of the difference for other use cases.

(closes issue ASTERISK-22084)
Review: https://reviewboard.asterisk.org/r/2726/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-13 15:27:32 +00:00
John Bigelow
d195728541 Add test suite events for when contacts are added or removed from an AOR
These are needed by the pjsip inbound registration test suite tests.

(issue ASTERISK-21833)
(issue ASTERISK-21834)
(issue ASTERISK-21835)
(issue ASTERISK-21837)

Review: https://reviewboard.asterisk.org/r/2700/
Review: https://reviewboard.asterisk.org/r/2739/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-12 22:05:18 +00:00
Jonathan Rose
6fe21ef48e bridge_channel: Support the lonely flag and make ARI use it.
The lonely flag is an optional flag for bridge channels that will
make them leave a bridge when a channel leaves if only lonely
channels are in the bridge at that point. This is useful for things
like ending recording and playback channels when they cease to be
interacting with other channels in the bridge.

(closes issue ASTERISK-22117)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2721/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-09 17:22:28 +00:00
David M. Lee
c790848794 ARI: Add recording controls
This patch implements the controls from ARI recordings. The controls
are:

 * DELETE /recordings/live/{recordingName} - stop recording and
   discard it
 * POST /recordings/live/{recordingName}/stop - stop recording
 * POST /recordings/live/{recordingName}/pause - pause recording
 * POST /recordings/live/{recordingName}/unpause - resume recording
 * POST /recordings/live/{recordingName}/mute - mute recording (record
   silence to the file)
 * POST /recordings/live/{recordingName}/unmute - unmute recording.

Since this underlying functionality did not already exist, is was
added to app.c by a set of control frames, similar to how playback
control works. The pause/mute control frames are toggles, even though
the ARI controls are idempotent, to be consistent with the playback
control frames.

(closes issue ASTERISK-22181)
Review: https://reviewboard.asterisk.org/r/2697/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-06 14:44:45 +00:00
Kinsey Moore
dfa5be76d4 Expose res_pjsip threadpool options
Expose initial size, automatic increment, maximum size, and idle
timeout as configurable parameters for the res_pjsip thread pool.

Review: https://reviewboard.asterisk.org/r/2704/
(closes issue ASTERISK-22143)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-06 13:08:13 +00:00
Joshua Colp
5b3441ae55 Fix crash in res_pjsip_outbound_registration when the remote server can not be resolved.
This crash was caused by decrementing the reference count of a newly created message when
it should not be. This change fixes that but also fixes all other cases where this was
incorrectly done.

(closes issue ASTERISK-22188)
Reported by: Kinsey Moore


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-06 12:39:27 +00:00
Matthew Jordan
80c9ad102e Add AMI registration events for PJSIP outbound registration attempts
This patch adds AMI events whenever an outbound registration attempt succeeds
or fails from res_pjsip_outbound_registration. This brings it inline with
the existing SIP channel driver and IAX channel driver.

Review: https://reviewboard.asterisk.org/r/2729/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-05 19:01:45 +00:00
Jonathan Rose
e47794ead1 ARI: bridges/{bridgeID}/addChannel: add roles parameter
Roles are now cleared with each entry into a bridge with addChannel.
If the roles parameter is present, the role specified will be applied
to all channels being added with the addChannel command.

(closes issue ASTERISK-21973)
Reported by: Matt Jordan
https://reviewboard.asterisk.org/r/2691/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-05 16:59:13 +00:00
Jonathan Rose
98b02d98f3 res_parking: Unit tests
Adds the following unit tests:
* create_lot: tests adding and removal of a new parking lot (baseline)
* park_extensions: creates a parking lot that registers extensions and
      then confirms that all of the expected extensions exist
* extensions_conflicts: creates numerous parking lots to test that
      extension conflicts in parking lots result in parking lot
      creation failing
* dynamic_parking_variables: Tests that the creation of dynamic
      parking lots respects the related channel variables set on the
      channel that requests them.
* park_call: Tests adding a channel to a parking lot's holding bridge
      by standard parking functions.
* retrieve_call: Tests pulling a channel out of a parking lot's
      holding bridge via parked call retrieval functions.

(closes issue ASTERISK-22138)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2714/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-05 16:00:01 +00:00
David M. Lee
357b275239 Fix res_ari_asterisk load issue
The new res_ari_asterisk.so module presents several config options
from asterisk main. Unfortunately, they aren't exported, so the module
won't load on Linux.

This patch renames the variables, adding the ast_ prefix so they will
be exported.

Review: https://reviewboard.asterisk.org/r/2737


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-05 14:35:00 +00:00
Mark Michelson
11d2993426 Get the SNMP code to compile.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 15:01:37 +00:00
David M. Lee
5114e4fc0b ARI - GET /ari/asterisk/info
This patch adds basic system information access to ARI.

The results are roughly what you get from 'core show settings', with a
few minor differences.

 * Data is structured, with 'build', 'system', 'config' and 'status'
   sub-objects.
 * Each sub-object is selectable, using the ?only= parameter. A comma
   separated list can be provided to select multiple sections.
 * A few config options are numeric, for which 0 means 'unlimited'.
   Instead of having a special interpretation of those fields, they
   are simply omitted if they're 0.
 * The information is limited to what might be useful to building
   external applications.

(closes issue ASTERISK-21575)
Review: https://reviewboard.asterisk.org/r/2702/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 14:46:21 +00:00
David M. Lee
537ecebd2d ARI - implement allowMultiple for parameters
Swagger allows parameters to be specified as 'allowMultiple', meaning
that the parameter may be specified as a comma separated list of
values.

I had written some of the API docs using that, but promptly forgot
about implementing it. This patch finally fills in that gap.

The codegen template was updated to represent 'allowMultiple' fields
as array/size fields in the _args structs. It also parses the comma
separated list using ast_app_separate_args(), so quoted strings in the
argument will be handled properly.

Review: https://reviewboard.asterisk.org/r/2698/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 14:36:32 +00:00
David M. Lee
10c91bc96e Address JSON thread safety issues.
In tracking down some unit tests failures, I ended up reading the fine
print[1] regarding Jansson's thread safety.

In short:
 1. Ref-counting is non-atomic.
 2. json_dumps() and friends are not thread safe.

This patch adds locking where necessary to our ast_json_* wrapper API,
with documentation in json.h describing the thread safety limitations of
the API.

 [1]: http://www.digip.org/jansson/doc/2.4/portability.html#thread-safety

Review: https://reviewboard.asterisk.org/r/2716/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 14:27:35 +00:00
Mark Michelson
328e99f41d Make a couple of changes to help AMI events to be more clear in what is occurring.
* BridgeEnter now contains the unique ID of the channel that is to be swapped out, if applicable.
* There is a ParkedCallSwap event that is sent when a parked channel has a new channel take its place.

(closes issue ASTERISK-22193)
reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/2712



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 14:13:04 +00:00
Kinsey Moore
41cd06e03f Add CLI/AMI commands to force chan_pjsip actions
For chan_pjsip, this introduces CLI/AMI remote unregistration commands,
reworks CLI syntax for sending NOTIFYs, adds AMI qualification support,
and adds documentation for PJSIPNotify.

This also fixes two refcounting bugs in the outbound registration code.

Review: https://reviewboard.asterisk.org/r/2695/
(closes issue ASTERISK-21939)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 12:40:03 +00:00
Matthew Jordan
38236e54a8 Remove dead code from features.c; refactor pickup code into pickup.c
This patch does the following:
 * It moves the pickup code out of features.c and into pickup.c
 * It removes the vast majority of dead code out of features.c. In particular,
   this includes the parking code.

(issue ASTERISK-22134)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-02 02:32:44 +00:00
Joshua Colp
63a229e369 Fix a crash due to performing full URI validation on a contact which only contains '*'.
(closes issue AST-1198)
Reported by: John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 23:38:00 +00:00
Matthew Jordan
5c4b482471 Support externally initiated parking requests; remove some dead code
This patch does the following:
 * It adds support for externally initiated parking requests. In particular,
   chan_skinny has a protocol level message that initiates a call park.
   This patch now supports that option, as well as the protocol specific
   mechanisms in chan_dahdi/sig_analog and chan_mgcp.
 * A parking bridge features virtual table has been added that provides
   access to the parking functionality that the Bridging API needs. This
   includes requests to park an entire 'call' (with little or no additional
   information, thank you chan_skinny), perform a blind transfer to a parking
   extension, determine if an extension is a parking extension, as well as the
   actual "do the parking" request from the Bridging API.
 * Refactoring in chan_mgcp, chan_skinny, and chan_dahdi to make use of the new
   functions
 * The removal of some - but not all - dead parking code from features.c

This also fixed blind transferring a multi-party bridge to a parking lot (which
was implemented, but had at least one code path where using the parking features
kK might not have worked)

Review: https://reviewboard.asterisk.org/r/2710

(closes issue ASTERISK-22134)
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 20:55:17 +00:00
Kinsey Moore
03090a88ba Fix documentation replication issues
This prevents XML documentation duplication by expanding channel and
bridge snapshot tags into channel and bridge snapshot parameter sets
with a given prefix or defaulting to no prefix. This also prevents
documentation from becoming fractured and out of date by keeping all
variations of the documentation in template form such that it only
needs to be updated once and keeps maintenance to a minimum.

Review: https://reviewboard.asterisk.org/r/2708/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 17:07:52 +00:00
David M. Lee
88d6c366d1 Fixed compile errors introduced in r395954.
Just a merge error due to a file rename. Grrr...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 15:31:03 +00:00
David M. Lee
e1b959ccbb Split caching out from the stasis_caching_topic.
In working with res_stasis, I discovered a significant limitation to
the current structure of stasis_caching_topics: you cannot subscribe
to cache updates for a single channel/bridge/endpoint/etc.

To address this, this patch splits the cache away from the
stasis_caching_topic, making it a first class object. The stasis_cache
object is shared amongst individual stasis_caching_topics that are
created per channel/endpoint/etc. These are still forwarded to global
whatever_all_cached topics, so their use from most of the code does
not change.

In making these changes, I noticed that we frequently used a similar
pattern for bridges, endpoints and channels:

     single_topic  ---------------->  all_topic
           ^
           |
     single_topic_cached  ----+---->  all_topic_cached
                              |
                              +---->  cache

This pattern was extracted as the 'Stasis Caching Pattern', defined in
stasis_caching_pattern.h. This avoids a lot of duplicate code between
the different domain objects.

Since the cache is now disassociated from its upstream caching topics,
this also necessitated a change to how the 'guaranteed' flag worked
for retrieving from a cache. The code for handling the caching
guarantee was extracted into a 'stasis_topic_wait' function, which
works for any stasis_topic.

(closes issue ASTERISK-22002)
Review: https://reviewboard.asterisk.org/r/2672/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01 13:49:34 +00:00