8 files changed, 638 insertions, 0 deletions
diff --git a/res/stasis_json/resource_asterisk.h b/res/stasis_json/resource_asterisk.h
new file mode 100644
index 000000000..5a717d005
--- /dev/null
+++ b/res/stasis_json/resource_asterisk.h
@@ -0,0 +1,46 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012 - 2013, Digium, Inc.
+ *
+ * David M. Lee, II <dlee@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Generated file - declares stubs to be implemented in
+ * res/stasis_json/resource_asterisk.c
+ *
+ * Asterisk resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * !!!!!                               DO NOT EDIT                        !!!!!
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * This file is generated by a mustache template. Please see the original
+ * template in rest-api-templates/stasis_http_resource.h.mustache
+ */
+
+#ifndef _ASTERISK_RESOURCE_ASTERISK_H
+#define _ASTERISK_RESOURCE_ASTERISK_H
+
+/*
+ * JSON models
+ *
+ * AsteriskInfo
+ */
+
+#endif /* _ASTERISK_RESOURCE_ASTERISK_H */
diff --git a/res/stasis_json/resource_bridges.h b/res/stasis_json/resource_bridges.h
new file mode 100644
index 000000000..cf2d03dc7
--- /dev/null
+++ b/res/stasis_json/resource_bridges.h
@@ -0,0 +1,48 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012 - 2013, Digium, Inc.
+ *
+ * David M. Lee, II <dlee@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Generated file - declares stubs to be implemented in
+ * res/stasis_json/resource_bridges.c
+ *
+ * Bridge resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * !!!!!                               DO NOT EDIT                        !!!!!
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * This file is generated by a mustache template. Please see the original
+ * template in rest-api-templates/stasis_http_resource.h.mustache
+ */
+
+#ifndef _ASTERISK_RESOURCE_BRIDGES_H
+#define _ASTERISK_RESOURCE_BRIDGES_H
+
+/*
+ * JSON models
+ *
+ * Bridge
+ * - channels: List[string] (required)
+ * - bridgeType: string (required)
+ */
+
+#endif /* _ASTERISK_RESOURCE_BRIDGES_H */
diff --git a/res/stasis_json/resource_channels.h b/res/stasis_json/resource_channels.h
new file mode 100644
index 000000000..45e1031fc
--- /dev/null
+++ b/res/stasis_json/resource_channels.h
@@ -0,0 +1,70 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012 - 2013, Digium, Inc.
+ *
+ * David M. Lee, II <dlee@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Generated file - declares stubs to be implemented in
+ * res/stasis_json/resource_channels.c
+ *
+ * Channel resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * !!!!!                               DO NOT EDIT                        !!!!!
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * This file is generated by a mustache template. Please see the original
+ * template in rest-api-templates/stasis_http_resource.h.mustache
+ */
+
+#ifndef _ASTERISK_RESOURCE_CHANNELS_H
+#define _ASTERISK_RESOURCE_CHANNELS_H
+
+/*
+ * JSON models
+ *
+ * Originated
+ * DialplanCEP
+ * - priority: long (required)
+ * - exten: string (required)
+ * - context: string (required)
+ * Channel
+ * - accountcode: string (required)
+ * - linkedid: string (required)
+ * - name: string (required)
+ * - userfield: string (required)
+ * - caller: CallerID (required)
+ * - creationtime: Date (required)
+ * - state: string (required)
+ * - parkinglot: string (required)
+ * - peeraccount: string (required)
+ * - appl: string (required)
+ * - connected: CallerID (required)
+ * - uniqueid: string (required)
+ * - hangupsource: string (required)
+ * - dialplan: DialplanCEP (required)
+ * - data: string (required)
+ * CallerID
+ * - name: string (required)
+ * - number: string (required)
+ * Dialed
+ */
+
+#endif /* _ASTERISK_RESOURCE_CHANNELS_H */
diff --git a/res/stasis_json/resource_endpoints.h b/res/stasis_json/resource_endpoints.h
new file mode 100644
index 000000000..7f2e4233c
--- /dev/null
+++ b/res/stasis_json/resource_endpoints.h
@@ -0,0 +1,48 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012 - 2013, Digium, Inc.
+ *
+ * David M. Lee, II <dlee@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Generated file - declares stubs to be implemented in
+ * res/stasis_json/resource_endpoints.c
+ *
+ * Endpoint resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * !!!!!                               DO NOT EDIT                        !!!!!
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * This file is generated by a mustache template. Please see the original
+ * template in rest-api-templates/stasis_http_resource.h.mustache
+ */
+
+#ifndef _ASTERISK_RESOURCE_ENDPOINTS_H
+#define _ASTERISK_RESOURCE_ENDPOINTS_H
+
+/*
+ * JSON models
+ *
+ * Endpoint
+ * - resource: string (required)
+ * - technology: string (required)
+ */
+
+#endif /* _ASTERISK_RESOURCE_ENDPOINTS_H */
diff --git a/res/stasis_json/resource_events.h b/res/stasis_json/resource_events.h
new file mode 100644
index 000000000..bd1c3263b
--- /dev/null
+++ b/res/stasis_json/resource_events.h
@@ -0,0 +1,275 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012 - 2013, Digium, Inc.
+ *
+ * David M. Lee, II <dlee@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Generated file - declares stubs to be implemented in
+ * res/stasis_json/resource_events.c
+ *
+ * WebSocket resource
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * !!!!!                               DO NOT EDIT                        !!!!!
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * This file is generated by a mustache template. Please see the original
+ * template in rest-api-templates/stasis_http_resource.h.mustache
+ */
+
+#ifndef _ASTERISK_RESOURCE_EVENTS_H
+#define _ASTERISK_RESOURCE_EVENTS_H
+
+struct ast_channel_snapshot;
+
+/*!
+ * \brief Some part of channel state changed.
+ *
+ * \param channel The channel to be used to generate this event
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_snapshot_create(
+	struct ast_channel_snapshot *channel_snapshot
+	);
+
+/*!
+ * \brief Notification that a channel has been destroyed.
+ *
+ * \param channel The channel to be used to generate this event
+ * \param blob JSON blob containing the following parameters:
+ * - cause: integer - Integer representation of the cause of the hangup (required)
+ * - cause_txt: string - Text representation of the cause of the hangup (required)
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_destroyed_create(
+	struct ast_channel_snapshot *channel_snapshot,
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief Channel changed Caller ID.
+ *
+ * \param channel The channel that changed Caller ID.
+ * \param blob JSON blob containing the following parameters:
+ * - caller_presentation_txt: string - The text representation of the Caller Presentation value. (required)
+ * - caller_presentation: integer - The integer representation of the Caller Presentation value. (required)
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_caller_id_create(
+	struct ast_channel_snapshot *channel_snapshot,
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief A hangup was requested on the channel.
+ *
+ * \param channel The channel on which the hangup was requested.
+ * \param blob JSON blob containing the following parameters:
+ * - soft: boolean - Whether the hangup request was a soft hangup request.
+ * - cause: integer - Integer representation of the cause of the hangup.
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_hangup_request_create(
+	struct ast_channel_snapshot *channel_snapshot,
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief Notification that another WebSocket has taken over for an application.
+ *
+ * \param blob JSON blob containing the following parameters:
+ * - application: string  (required)
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_application_replaced_create(
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief Channel variable changed.
+ *
+ * \param channel The channel on which the variable was set.
+ * \param blob JSON blob containing the following parameters:
+ * - variable: string - The variable that changed. (required)
+ * - value: string - The new value of the variable. (required)
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_varset_create(
+	struct ast_channel_snapshot *channel_snapshot,
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief User-generated event with additional user-defined fields in the object.
+ *
+ * \param channel The channel that signaled the user event.
+ * \param blob JSON blob containing the following parameters:
+ * - eventname: string - The name of the user event. (required)
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_userevent_create(
+	struct ast_channel_snapshot *channel_snapshot,
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief Notification that a channel has been created.
+ *
+ * \param channel The channel to be used to generate this event
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_created_create(
+	struct ast_channel_snapshot *channel_snapshot
+	);
+
+/*!
+ * \brief Notification that a channel has entered a Stasis appliction.
+ *
+ * \param channel The channel to be used to generate this event
+ * \param blob JSON blob containing the following parameters:
+ * - args: List[string] - Arguments to the application (required)
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_stasis_start_create(
+	struct ast_channel_snapshot *channel_snapshot,
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief Channel changed location in the dialplan.
+ *
+ * \param channel The channel that changed dialplan location.
+ * \param blob JSON blob containing the following parameters:
+ * - application: string - The application that the channel is currently in. (required)
+ * - application_data: string - The data that was passed to the application when it was invoked. (required)
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_dialplan_create(
+	struct ast_channel_snapshot *channel_snapshot,
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief Notification of a channel's state change.
+ *
+ * \param channel The channel to be used to generate this event
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_state_change_create(
+	struct ast_channel_snapshot *channel_snapshot
+	);
+
+/*!
+ * \brief DTMF received on a channel.
+ *
+ * \param channel The channel on which DTMF was received
+ * \param blob JSON blob containing the following parameters:
+ * - digit: string - DTMF digit received (0-9, A-E, # or *) (required)
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_channel_dtmf_received_create(
+	struct ast_channel_snapshot *channel_snapshot,
+	struct ast_json *blob
+	);
+
+/*!
+ * \brief Notification that a channel has left a Stasis appliction.
+ *
+ * \param channel The channel to be used to generate this event
+ *
+ * \retval NULL on error
+ * \retval JSON (ast_json) describing the event
+ */
+struct ast_json *stasis_json_event_stasis_end_create(
+	struct ast_channel_snapshot *channel_snapshot
+	);
+
+/*
+ * JSON models
+ *
+ * ChannelSnapshot
+ * ChannelDestroyed
+ * - cause: integer (required)
+ * - cause_txt: string (required)
+ * ChannelCallerId
+ * - caller_presentation_txt: string (required)
+ * - caller_presentation: integer (required)
+ * ChannelHangupRequest
+ * - soft: boolean
+ * - cause: integer
+ * ApplicationReplaced
+ * - application: string (required)
+ * ChannelVarset
+ * - variable: string (required)
+ * - value: string (required)
+ * ChannelUserevent
+ * - eventname: string (required)
+ * ChannelCreated
+ * StasisStart
+ * - args: List[string] (required)
+ * ChannelDialplan
+ * - application: string (required)
+ * - application_data: string (required)
+ * ChannelStateChange
+ * ChannelDtmfReceived
+ * - digit: string (required)
+ * Event
+ * - channel_created: ChannelCreated
+ * - channel_destroyed: ChannelDestroyed
+ * - channel_dialplan: ChannelDialplan
+ * - channel_varset: ChannelVarset
+ * - application_replaced: ApplicationReplaced
+ * - channel_state_change: ChannelStateChange
+ * - stasis_start: StasisStart
+ * - application: string (required)
+ * - channel_hangup_request: ChannelHangupRequest
+ * - channel_userevent: ChannelUserevent
+ * - channel_snapshot: ChannelSnapshot
+ * - channel_dtmf_received: ChannelDtmfReceived
+ * - channel_caller_id: ChannelCallerId
+ * - stasis_end: StasisEnd
+ * StasisEnd
+ */
+
+#endif /* _ASTERISK_RESOURCE_EVENTS_H */
diff --git a/res/stasis_json/resource_playback.h b/res/stasis_json/resource_playback.h
new file mode 100644
index 000000000..e84e6de0d
--- /dev/null
+++ b/res/stasis_json/resource_playback.h
@@ -0,0 +1,47 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012 - 2013, Digium, Inc.
+ *
+ * David M. Lee, II <dlee@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Generated file - declares stubs to be implemented in
+ * res/stasis_json/resource_playback.c
+ *
+ * Playback control resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * !!!!!                               DO NOT EDIT                        !!!!!
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * This file is generated by a mustache template. Please see the original
+ * template in rest-api-templates/stasis_http_resource.h.mustache
+ */
+
+#ifndef _ASTERISK_RESOURCE_PLAYBACK_H
+#define _ASTERISK_RESOURCE_PLAYBACK_H
+
+/*
+ * JSON models
+ *
+ * Playback
+ * - id: string (required)
+ */
+
+#endif /* _ASTERISK_RESOURCE_PLAYBACK_H */
diff --git a/res/stasis_json/resource_recordings.h b/res/stasis_json/resource_recordings.h
new file mode 100644
index 000000000..b460fb769
--- /dev/null
+++ b/res/stasis_json/resource_recordings.h
@@ -0,0 +1,54 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012 - 2013, Digium, Inc.
+ *
+ * David M. Lee, II <dlee@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Generated file - declares stubs to be implemented in
+ * res/stasis_json/resource_recordings.c
+ *
+ * Recording resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * !!!!!                               DO NOT EDIT                        !!!!!
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * This file is generated by a mustache template. Please see the original
+ * template in rest-api-templates/stasis_http_resource.h.mustache
+ */
+
+#ifndef _ASTERISK_RESOURCE_RECORDINGS_H
+#define _ASTERISK_RESOURCE_RECORDINGS_H
+
+/*
+ * JSON models
+ *
+ * Recording
+ * - id: string (required)
+ * StoredRecording
+ * - durationSeconds: int
+ * - time: Date
+ * - id: string (required)
+ * - formats: List[string] (required)
+ * LiveRecording
+ * - id: string (required)
+ */
+
+#endif /* _ASTERISK_RESOURCE_RECORDINGS_H */
diff --git a/res/stasis_json/resource_sounds.h b/res/stasis_json/resource_sounds.h
new file mode 100644
index 000000000..306e2e4a8
--- /dev/null
+++ b/res/stasis_json/resource_sounds.h
@@ -0,0 +1,50 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2012 - 2013, Digium, Inc.
+ *
+ * David M. Lee, II <dlee@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Generated file - declares stubs to be implemented in
+ * res/stasis_json/resource_sounds.c
+ *
+ * Sound resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * !!!!!                               DO NOT EDIT                        !!!!!
+ * !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
+ * This file is generated by a mustache template. Please see the original
+ * template in rest-api-templates/stasis_http_resource.h.mustache
+ */
+
+#ifndef _ASTERISK_RESOURCE_SOUNDS_H
+#define _ASTERISK_RESOURCE_SOUNDS_H
+
+/*
+ * JSON models
+ *
+ * Sound
+ * - lang: string (required)
+ * - text: string
+ * - id: string (required)
+ * - formats: List[string] (required)
+ */
+
+#endif /* _ASTERISK_RESOURCE_SOUNDS_H */