This patch adds a RESTful HTTP interface to Asterisk.

The API itself is documented using Swagger, a lightweight mechanism for
documenting RESTful API's using JSON. This allows us to use swagger-ui
to provide executable documentation for the API, generate client
bindings in different languages, and generate a lot of the boilerplate
code for implementing the RESTful bindings. The API docs live in the
rest-api/ directory.

The RESTful bindings are generated from the Swagger API docs using a set
of Mustache templates.  The code generator is written in Python, and
uses Pystache. Pystache has no dependencies, and be installed easily
using pip. Code generation code lives in rest-api-templates/.

The generated code reduces a lot of boilerplate when it comes to
handling HTTP requests. It also helps us have greater consistency in the
REST API.

(closes issue ASTERISK-20891)
Review: https://reviewboard.asterisk.org/r/2376/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386232 65c4cc65-6c06-0410-ace0-fbb531ad65f3

16 files changed, 1524 insertions, 0 deletions

diff --git a/res/stasis_http/resource_asterisk.c b/res/stasis_http/resource_asterisk.c
new file mode 100644
index 000000000..b5e8b0f54
--- /dev/null
+++ b/res/stasis_http/resource_asterisk.c
@@ -0,0 +1,39 @@
+/* -*- C -*-
+ * 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 Implementation for stasis-http stubs.
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "resource_asterisk.h"
+
+void stasis_http_get_asterisk_info(struct ast_variable *headers, struct ast_get_asterisk_info_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_asterisk_info\n");
+}
diff --git a/res/stasis_http/resource_asterisk.h b/res/stasis_http/resource_asterisk.h
new file mode 100644
index 000000000..0d373ccb2
--- /dev/null
+++ b/res/stasis_http/resource_asterisk.h
@@ -0,0 +1,56 @@
+/*
+ * 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_http/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
+
+#include "asterisk/stasis_http.h"
+
+/*! \brief Argument struct for stasis_http_get_asterisk_info() */
+struct ast_get_asterisk_info_args {
+	/*! \brief Filter information returned */
+	const char *only;
+};
+/*!
+ * \brief Gets Asterisk system information.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_asterisk_info(struct ast_variable *headers, struct ast_get_asterisk_info_args *args, struct stasis_http_response *response);
+
+#endif /* _ASTERISK_RESOURCE_ASTERISK_H */
diff --git a/res/stasis_http/resource_bridges.c b/res/stasis_http/resource_bridges.c
new file mode 100644
index 000000000..ca48ee725
--- /dev/null
+++ b/res/stasis_http/resource_bridges.c
@@ -0,0 +1,63 @@
+/* -*- C -*-
+ * 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 Implementation for stasis-http stubs.
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "resource_bridges.h"
+
+void stasis_http_add_channel_to_bridge(struct ast_variable *headers, struct ast_add_channel_to_bridge_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_add_channel_to_bridge\n");
+}
+void stasis_http_remove_channel_from_bridge(struct ast_variable *headers, struct ast_remove_channel_from_bridge_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_remove_channel_from_bridge\n");
+}
+void stasis_http_record_bridge(struct ast_variable *headers, struct ast_record_bridge_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_record_bridge\n");
+}
+void stasis_http_get_bridge(struct ast_variable *headers, struct ast_get_bridge_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_bridge\n");
+}
+void stasis_http_delete_bridge(struct ast_variable *headers, struct ast_delete_bridge_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_delete_bridge\n");
+}
+void stasis_http_get_bridges(struct ast_variable *headers, struct ast_get_bridges_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_bridges\n");
+}
+void stasis_http_new_bridge(struct ast_variable *headers, struct ast_new_bridge_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_new_bridge\n");
+}
diff --git a/res/stasis_http/resource_bridges.h b/res/stasis_http/resource_bridges.h
new file mode 100644
index 000000000..db6db205b
--- /dev/null
+++ b/res/stasis_http/resource_bridges.h
@@ -0,0 +1,154 @@
+/*
+ * 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_http/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
+
+#include "asterisk/stasis_http.h"
+
+/*! \brief Argument struct for stasis_http_get_bridges() */
+struct ast_get_bridges_args {
+};
+/*!
+ * \brief List active bridges.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_bridges(struct ast_variable *headers, struct ast_get_bridges_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_new_bridge() */
+struct ast_new_bridge_args {
+	/*! \brief Type of bridge to create. */
+	const char *type;
+};
+/*!
+ * \brief Create a new bridge.
+ *
+ * This bridge persists until it has been shut down, or Asterisk has been shut down.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_new_bridge(struct ast_variable *headers, struct ast_new_bridge_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_get_bridge() */
+struct ast_get_bridge_args {
+	/*! \brief Bridge's id */
+	const char *bridge_id;
+};
+/*!
+ * \brief Get bridge details.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_bridge(struct ast_variable *headers, struct ast_get_bridge_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_delete_bridge() */
+struct ast_delete_bridge_args {
+	/*! \brief Bridge's id */
+	const char *bridge_id;
+};
+/*!
+ * \brief Shut down a bridge bridge.
+ *
+ * If any channels are in this bridge, they will be removed and resume whatever they were doing beforehand.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_delete_bridge(struct ast_variable *headers, struct ast_delete_bridge_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_add_channel_to_bridge() */
+struct ast_add_channel_to_bridge_args {
+	/*! \brief Bridge's id */
+	const char *bridge_id;
+	/*! \brief Channel's id */
+	const char *channel;
+};
+/*!
+ * \brief Add a channel to a bridge.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_add_channel_to_bridge(struct ast_variable *headers, struct ast_add_channel_to_bridge_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_remove_channel_from_bridge() */
+struct ast_remove_channel_from_bridge_args {
+	/*! \brief Bridge's id */
+	const char *bridge_id;
+	/*! \brief Channel's id */
+	const char *channel;
+};
+/*!
+ * \brief Remove a channel from a bridge.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_remove_channel_from_bridge(struct ast_variable *headers, struct ast_remove_channel_from_bridge_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_record_bridge() */
+struct ast_record_bridge_args {
+	/*! \brief Bridge's id */
+	const char *bridge_id;
+	/*! \brief Recording's filename */
+	const char *name;
+	/*! \brief Maximum duration of the recording, in seconds. 0 for no limit. */
+	int max_duration_seconds;
+	/*! \brief Maximum duration of silence, in seconds. 0 for no limit. */
+	int max_silence_seconds;
+	/*! \brief If true, and recording already exists, append to recording. */
+	int append;
+	/*! \brief Play beep when recording begins */
+	int beep;
+	/*! \brief DTMF input to terminate recording. */
+	const char *terminate_on;
+};
+/*!
+ * \brief Start a recording.
+ *
+ * This records the mixed audio from all channels participating in this bridge.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_record_bridge(struct ast_variable *headers, struct ast_record_bridge_args *args, struct stasis_http_response *response);
+
+#endif /* _ASTERISK_RESOURCE_BRIDGES_H */
diff --git a/res/stasis_http/resource_channels.c b/res/stasis_http/resource_channels.c
new file mode 100644
index 000000000..3cc97c511
--- /dev/null
+++ b/res/stasis_http/resource_channels.c
@@ -0,0 +1,251 @@
+/* -*- C -*-
+ * 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 Implementation for stasis-http stubs.
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/stasis_app.h"
+#include "asterisk/stasis_channels.h"
+#include "resource_channels.h"
+
+/*!
+ * \brief Finds the control object for a channel, filling the response with an
+ * error, if appropriate.
+ * \param[out] response Response to fill with an error if control is not found.
+ * \param channel_id ID of the channel to lookup.
+ * \return Channel control object.
+ * \return \c NULL if control object does not exist.
+ */
+static struct stasis_app_control *find_control(
+	struct stasis_http_response *response,
+	const char *channel_id)
+{
+	RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
+
+	ast_assert(response != NULL);
+
+	control = stasis_app_control_find_by_channel_id(channel_id);
+	if (control == NULL) {
+		/* Distinguish between 404 and 409 errors */
+		RAII_VAR(struct ast_channel *, chan, NULL, ao2_cleanup);
+		chan = ast_channel_get_by_name(channel_id);
+		if (chan == NULL) {
+			stasis_http_response_error(response, 404, "Not Found",
+				   "Channel not found");
+			return NULL;
+		}
+
+		stasis_http_response_error(response, 409, "Conflict",
+			   "Channel not in Stasis application");
+		return NULL;
+	}
+
+	ao2_ref(control, +1);
+	return control;
+}
+
+void stasis_http_dial(struct ast_variable *headers, struct ast_dial_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_dial\n");
+}
+
+void stasis_http_continue_in_dialplan(
+	struct ast_variable *headers,
+	struct ast_continue_in_dialplan_args *args,
+	struct stasis_http_response *response)
+{
+	RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
+
+	ast_assert(response != NULL);
+
+	control = find_control(response, args->channel_id);
+	if (control == NULL) {
+		return;
+	}
+
+	stasis_app_control_continue(control);
+	stasis_http_response_no_content(response);
+}
+
+void stasis_http_answer_channel(struct ast_variable *headers,
+				struct ast_answer_channel_args *args,
+				struct stasis_http_response *response)
+{
+	RAII_VAR(struct stasis_app_control *, control, NULL, ao2_cleanup);
+
+	control = find_control(response, args->channel_id);
+	if (control == NULL) {
+		return;
+	}
+
+	if (stasis_app_control_answer(control) != 0) {
+		stasis_http_response_error(
+			response, 500, "Internal Server Error",
+			"Failed to answer channel");
+		return;
+	}
+
+	stasis_http_response_no_content(response);
+}
+
+void stasis_http_mute_channel(struct ast_variable *headers, struct ast_mute_channel_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_mute_channel\n");
+}
+void stasis_http_unmute_channel(struct ast_variable *headers, struct ast_unmute_channel_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_unmute_channel\n");
+}
+void stasis_http_hold_channel(struct ast_variable *headers, struct ast_hold_channel_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_hold_channel\n");
+}
+void stasis_http_unhold_channel(struct ast_variable *headers, struct ast_unhold_channel_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_unhold_channel\n");
+}
+void stasis_http_play_on_channel(struct ast_variable *headers, struct ast_play_on_channel_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_play_on_channel\n");
+}
+void stasis_http_record_channel(struct ast_variable *headers, struct ast_record_channel_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_record_channel\n");
+}
+void stasis_http_get_channel(struct ast_variable *headers,
+			     struct ast_get_channel_args *args,
+			     struct stasis_http_response *response)
+{
+	RAII_VAR(struct stasis_caching_topic *, caching_topic, NULL, ao2_cleanup);
+	RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
+	struct ast_channel_snapshot *snapshot;
+
+	caching_topic = ast_channel_topic_all_cached();
+	if (!caching_topic) {
+		stasis_http_response_error(
+			response, 500, "Internal Server Error",
+			"Message bus not initialized");
+		return;
+	}
+	ao2_ref(caching_topic, +1);
+
+	msg = stasis_cache_get(caching_topic, ast_channel_snapshot_type(),
+			       args->channel_id);
+	if (!msg) {
+		stasis_http_response_error(
+			response, 404, "Not Found",
+			"Channel not found");
+		return;
+	}
+
+	snapshot = stasis_message_data(msg);
+	ast_assert(snapshot != NULL);
+
+	stasis_http_response_ok(response,
+				ast_channel_snapshot_to_json(snapshot));
+}
+
+void stasis_http_delete_channel(struct ast_variable *headers,
+				struct ast_delete_channel_args *args,
+				struct stasis_http_response *response)
+{
+	RAII_VAR(struct ast_channel *, chan, NULL, ao2_cleanup);
+
+	chan = ast_channel_get_by_name(args->channel_id);
+	if (chan == NULL) {
+		stasis_http_response_error(
+			response, 404, "Not Found",
+			"Channel not found");
+		return;
+	}
+
+	ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
+
+	stasis_http_response_no_content(response);
+}
+
+void stasis_http_get_channels(struct ast_variable *headers,
+			      struct ast_get_channels_args *args,
+			      struct stasis_http_response *response)
+{
+	RAII_VAR(struct stasis_caching_topic *, caching_topic, NULL, ao2_cleanup);
+	RAII_VAR(struct ao2_container *, snapshots, NULL, ao2_cleanup);
+	RAII_VAR(struct ast_json *, json, NULL, ast_json_unref);
+	struct ao2_iterator i;
+	void *obj;
+
+	caching_topic = ast_channel_topic_all_cached();
+	if (!caching_topic) {
+		stasis_http_response_error(
+			response, 500, "Internal Server Error",
+			"Message bus not initialized");
+		return;
+	}
+	ao2_ref(caching_topic, +1);
+
+	snapshots = stasis_cache_dump(caching_topic, ast_channel_snapshot_type());
+	if (!snapshots) {
+		stasis_http_response_alloc_failed(response);
+		return;
+	}
+
+	json = ast_json_array_create();
+	if (!json) {
+		stasis_http_response_alloc_failed(response);
+		return;
+	}
+
+	i = ao2_iterator_init(snapshots, 0);
+	while ((obj = ao2_iterator_next(&i))) {
+		RAII_VAR(struct stasis_message *, msg, obj, ao2_cleanup);
+		struct ast_channel_snapshot *snapshot = stasis_message_data(msg);
+		int r = ast_json_array_append(
+			json, ast_channel_snapshot_to_json(snapshot));
+		if (r != 0) {
+			stasis_http_response_alloc_failed(response);
+			return;
+		}
+	}
+	ao2_iterator_destroy(&i);
+
+	stasis_http_response_ok(response, ast_json_ref(json));
+}
+
+void stasis_http_originate(struct ast_variable *headers,
+			   struct ast_originate_args *args,
+			   struct stasis_http_response *response)
+{
+	if (args->endpoint) {
+		ast_log(LOG_DEBUG, "Dialing specific endpoint %s\n", args->endpoint);
+	}
+
+	ast_log(LOG_DEBUG, "Dialing %s@%s\n", args->extension, args->context);
+	/* ast_pbx_outgoing_app - originates a channel, putting it into an application */
+}
diff --git a/res/stasis_http/resource_channels.h b/res/stasis_http/resource_channels.h
new file mode 100644
index 000000000..2c78589b7
--- /dev/null
+++ b/res/stasis_http/resource_channels.h
@@ -0,0 +1,244 @@
+/*
+ * 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_http/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
+
+#include "asterisk/stasis_http.h"
+
+/*! \brief Argument struct for stasis_http_get_channels() */
+struct ast_get_channels_args {
+};
+/*!
+ * \brief List active channels.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_channels(struct ast_variable *headers, struct ast_get_channels_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_originate() */
+struct ast_originate_args {
+	/*! \brief Endpoint to call. If not specified, originate is routed via dialplan */
+	const char *endpoint;
+	/*! \brief Extension to dial */
+	const char *extension;
+	/*! \brief When routing via dialplan, the context use. If omitted, uses 'default' */
+	const char *context;
+};
+/*!
+ * \brief Create a new channel (originate).
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_originate(struct ast_variable *headers, struct ast_originate_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_get_channel() */
+struct ast_get_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+};
+/*!
+ * \brief Channel details.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_channel(struct ast_variable *headers, struct ast_get_channel_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_delete_channel() */
+struct ast_delete_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+};
+/*!
+ * \brief Delete (i.e. hangup) a channel.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_delete_channel(struct ast_variable *headers, struct ast_delete_channel_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_dial() */
+struct ast_dial_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+	/*! \brief Endpoint to call. If not specified, dial is routed via dialplan */
+	const char *endpoint;
+	/*! \brief Extension to dial */
+	const char *extension;
+	/*! \brief When routing via dialplan, the context use. If omitted, uses 'default' */
+	const char *context;
+};
+/*!
+ * \brief Create a new channel (originate) and bridge to this channel.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_dial(struct ast_variable *headers, struct ast_dial_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_continue_in_dialplan() */
+struct ast_continue_in_dialplan_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+};
+/*!
+ * \brief Exit application; continue execution in the dialplan.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_continue_in_dialplan(struct ast_variable *headers, struct ast_continue_in_dialplan_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_answer_channel() */
+struct ast_answer_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+};
+/*!
+ * \brief Answer a channel.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_answer_channel(struct ast_variable *headers, struct ast_answer_channel_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_mute_channel() */
+struct ast_mute_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+	/*! \brief Direction in which to mute audio */
+	const char *direction;
+};
+/*!
+ * \brief Mute a channel.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_mute_channel(struct ast_variable *headers, struct ast_mute_channel_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_unmute_channel() */
+struct ast_unmute_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+	/*! \brief Direction in which to unmute audio */
+	const char *direction;
+};
+/*!
+ * \brief Unmute a channel.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_unmute_channel(struct ast_variable *headers, struct ast_unmute_channel_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_hold_channel() */
+struct ast_hold_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+};
+/*!
+ * \brief Hold a channel.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_hold_channel(struct ast_variable *headers, struct ast_hold_channel_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_unhold_channel() */
+struct ast_unhold_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+};
+/*!
+ * \brief Remove a channel from hold.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_unhold_channel(struct ast_variable *headers, struct ast_unhold_channel_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_play_on_channel() */
+struct ast_play_on_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+	/*! \brief Media's URI to play. */
+	const char *media;
+};
+/*!
+ * \brief Start playback of media.
+ *
+ * The media URI may be any of a number of URI's. You may use http: and https: URI's, as well as sound: and recording: URI's. This operation creates a playback resource that can be used to control the playback of media (pause, rewind, fast forward, etc.)
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_play_on_channel(struct ast_variable *headers, struct ast_play_on_channel_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_record_channel() */
+struct ast_record_channel_args {
+	/*! \brief Channel's id */
+	const char *channel_id;
+	/*! \brief Recording's filename */
+	const char *name;
+	/*! \brief Format to encode audio in */
+	const char *format;
+	/*! \brief Maximum duration of the recording, in seconds. 0 for no limit */
+	int max_duration_seconds;
+	/*! \brief Maximum duration of silence, in seconds. 0 for no limit */
+	int max_silence_seconds;
+	/*! \brief If true, and recording already exists, append to recording */
+	int append;
+	/*! \brief Play beep when recording begins */
+	int beep;
+	/*! \brief DTMF input to terminate recording */
+	const char *terminate_on;
+};
+/*!
+ * \brief Start a recording.
+ *
+ * Record audio from a channel. Note that this will not capture audio sent to the channel. The bridge itself has a record feature if that's what you want.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_record_channel(struct ast_variable *headers, struct ast_record_channel_args *args, struct stasis_http_response *response);
+
+#endif /* _ASTERISK_RESOURCE_CHANNELS_H */
diff --git a/res/stasis_http/resource_endpoints.c b/res/stasis_http/resource_endpoints.c
new file mode 100644
index 000000000..b2611bad8
--- /dev/null
+++ b/res/stasis_http/resource_endpoints.c
@@ -0,0 +1,43 @@
+/* -*- C -*-
+ * 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 Implementation for stasis-http stubs.
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "resource_endpoints.h"
+
+void stasis_http_get_endpoint(struct ast_variable *headers, struct ast_get_endpoint_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_endpoint\n");
+}
+void stasis_http_get_endpoints(struct ast_variable *headers, struct ast_get_endpoints_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_endpoints\n");
+}
diff --git a/res/stasis_http/resource_endpoints.h b/res/stasis_http/resource_endpoints.h
new file mode 100644
index 000000000..57f4b91ba
--- /dev/null
+++ b/res/stasis_http/resource_endpoints.h
@@ -0,0 +1,69 @@
+/*
+ * 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_http/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
+
+#include "asterisk/stasis_http.h"
+
+/*! \brief Argument struct for stasis_http_get_endpoints() */
+struct ast_get_endpoints_args {
+	/*! \brief Filter endpoints by type (sip,iax2,dhadi,...) */
+	const char *with_type;
+};
+/*!
+ * \brief List available endoints.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_endpoints(struct ast_variable *headers, struct ast_get_endpoints_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_get_endpoint() */
+struct ast_get_endpoint_args {
+	/*! \brief ID of the endpoint */
+	const char *endpoint_id;
+};
+/*!
+ * \brief Details for an endpoint.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_endpoint(struct ast_variable *headers, struct ast_get_endpoint_args *args, struct stasis_http_response *response);
+
+#endif /* _ASTERISK_RESOURCE_ENDPOINTS_H */
diff --git a/res/stasis_http/resource_events.c b/res/stasis_http/resource_events.c
new file mode 100644
index 000000000..34563fe6e
--- /dev/null
+++ b/res/stasis_http/resource_events.c
@@ -0,0 +1,40 @@
+/* -*- C -*-
+ * 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 Implementation for stasis-http stubs.
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "resource_events.h"
+
+void stasis_http_event_websocket(struct ast_variable *headers, struct ast_event_websocket_args *args, struct stasis_http_response *response)
+{
+	/* TODO: This should promote this socket to a websocket connection */
+	ast_log(LOG_ERROR, "TODO: stasis_http_event_websocket\n");
+}
diff --git a/res/stasis_http/resource_events.h b/res/stasis_http/resource_events.h
new file mode 100644
index 000000000..0f58476df
--- /dev/null
+++ b/res/stasis_http/resource_events.h
@@ -0,0 +1,58 @@
+/*
+ * 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_http/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
+
+#include "asterisk/stasis_http.h"
+
+/*! \brief Argument struct for stasis_http_event_websocket() */
+struct ast_event_websocket_args {
+	/*! \brief Comma seperated list of applications to subscribe to. */
+	const char *app;
+	/*! \brief RFC6455 header for upgrading a connection to a websocket. */
+	const char *upgrade;
+};
+/*!
+ * \brief WebSocket connection for events.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_event_websocket(struct ast_variable *headers, struct ast_event_websocket_args *args, struct stasis_http_response *response);
+
+#endif /* _ASTERISK_RESOURCE_EVENTS_H */
diff --git a/res/stasis_http/resource_playback.c b/res/stasis_http/resource_playback.c
new file mode 100644
index 000000000..99f2e09eb
--- /dev/null
+++ b/res/stasis_http/resource_playback.c
@@ -0,0 +1,43 @@
+/*
+ * 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 /api-docs/playback.{format} implementation- Playback control resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "resource_playback.h"
+
+void stasis_http_get_playback(struct ast_variable *headers, struct ast_get_playback_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_playback\n");
+}
+void stasis_http_stop_playback(struct ast_variable *headers, struct ast_stop_playback_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_stop_playback\n");
+}
+void stasis_http_control_playback(struct ast_variable *headers, struct ast_control_playback_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_control_playback\n");
+}
diff --git a/res/stasis_http/resource_playback.h b/res/stasis_http/resource_playback.h
new file mode 100644
index 000000000..36b05bc04
--- /dev/null
+++ b/res/stasis_http/resource_playback.h
@@ -0,0 +1,84 @@
+/*
+ * 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_http/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
+
+#include "asterisk/stasis_http.h"
+
+/*! \brief Argument struct for stasis_http_get_playback() */
+struct ast_get_playback_args {
+	/*! \brief Playback's id */
+	const char *playback_id;
+};
+/*!
+ * \brief Get a playback's details.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_playback(struct ast_variable *headers, struct ast_get_playback_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_stop_playback() */
+struct ast_stop_playback_args {
+	/*! \brief Playback's id */
+	const char *playback_id;
+};
+/*!
+ * \brief Stop a playback.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_stop_playback(struct ast_variable *headers, struct ast_stop_playback_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_control_playback() */
+struct ast_control_playback_args {
+	/*! \brief Playback's id */
+	const char *playback_id;
+	/*! \brief Operation to perform on the playback. */
+	const char *operation;
+};
+/*!
+ * \brief Get a playback's details.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_control_playback(struct ast_variable *headers, struct ast_control_playback_args *args, struct stasis_http_response *response);
+
+#endif /* _ASTERISK_RESOURCE_PLAYBACK_H */
diff --git a/res/stasis_http/resource_recordings.c b/res/stasis_http/resource_recordings.c
new file mode 100644
index 000000000..2400a6876
--- /dev/null
+++ b/res/stasis_http/resource_recordings.c
@@ -0,0 +1,79 @@
+/*
+ * 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 /api-docs/recordings.{format} implementation- Recording resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "resource_recordings.h"
+
+void stasis_http_get_recordings(struct ast_variable *headers, struct ast_get_recordings_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_recordings\n");
+}
+void stasis_http_get_stored_recordings(struct ast_variable *headers, struct ast_get_stored_recordings_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_stored_recordings\n");
+}
+void stasis_http_get_stored_recording(struct ast_variable *headers, struct ast_get_stored_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_stored_recording\n");
+}
+void stasis_http_delete_stored_recording(struct ast_variable *headers, struct ast_delete_stored_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_delete_stored_recording\n");
+}
+void stasis_http_get_live_recordings(struct ast_variable *headers, struct ast_get_live_recordings_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_live_recordings\n");
+}
+void stasis_http_get_live_recording(struct ast_variable *headers, struct ast_get_live_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_live_recording\n");
+}
+void stasis_http_cancel_recording(struct ast_variable *headers, struct ast_cancel_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_cancel_recording\n");
+}
+void stasis_http_stop_recording(struct ast_variable *headers, struct ast_stop_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_stop_recording\n");
+}
+void stasis_http_pause_recording(struct ast_variable *headers, struct ast_pause_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_pause_recording\n");
+}
+void stasis_http_unpause_recording(struct ast_variable *headers, struct ast_unpause_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_unpause_recording\n");
+}
+void stasis_http_mute_recording(struct ast_variable *headers, struct ast_mute_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_mute_recording\n");
+}
+void stasis_http_unmute_recording(struct ast_variable *headers, struct ast_unmute_recording_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_unmute_recording\n");
+}
diff --git a/res/stasis_http/resource_recordings.h b/res/stasis_http/resource_recordings.h
new file mode 100644
index 000000000..ee48e43b7
--- /dev/null
+++ b/res/stasis_http/resource_recordings.h
@@ -0,0 +1,193 @@
+/*
+ * 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_http/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
+
+#include "asterisk/stasis_http.h"
+
+/*! \brief Argument struct for stasis_http_get_recordings() */
+struct ast_get_recordings_args {
+};
+/*!
+ * \brief List all recordings.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_recordings(struct ast_variable *headers, struct ast_get_recordings_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_get_stored_recordings() */
+struct ast_get_stored_recordings_args {
+};
+/*!
+ * \brief List recordings that are complete.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_stored_recordings(struct ast_variable *headers, struct ast_get_stored_recordings_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_get_stored_recording() */
+struct ast_get_stored_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief Get a stored recording's details.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_stored_recording(struct ast_variable *headers, struct ast_get_stored_recording_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_delete_stored_recording() */
+struct ast_delete_stored_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief Delete a stored recording.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_delete_stored_recording(struct ast_variable *headers, struct ast_delete_stored_recording_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_get_live_recordings() */
+struct ast_get_live_recordings_args {
+};
+/*!
+ * \brief List libe recordings.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_live_recordings(struct ast_variable *headers, struct ast_get_live_recordings_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_get_live_recording() */
+struct ast_get_live_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief List live recordings.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_live_recording(struct ast_variable *headers, struct ast_get_live_recording_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_cancel_recording() */
+struct ast_cancel_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief Stop a live recording and discard it.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_cancel_recording(struct ast_variable *headers, struct ast_cancel_recording_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_stop_recording() */
+struct ast_stop_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief Stop a live recording and store it.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_stop_recording(struct ast_variable *headers, struct ast_stop_recording_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_pause_recording() */
+struct ast_pause_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief Pause a live recording.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_pause_recording(struct ast_variable *headers, struct ast_pause_recording_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_unpause_recording() */
+struct ast_unpause_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief Unpause a live recording.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_unpause_recording(struct ast_variable *headers, struct ast_unpause_recording_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_mute_recording() */
+struct ast_mute_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief Mute a live recording.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_mute_recording(struct ast_variable *headers, struct ast_mute_recording_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_unmute_recording() */
+struct ast_unmute_recording_args {
+	/*! \brief Recording's id */
+	const char *recording_id;
+};
+/*!
+ * \brief Unmute a live recording.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_unmute_recording(struct ast_variable *headers, struct ast_unmute_recording_args *args, struct stasis_http_response *response);
+
+#endif /* _ASTERISK_RESOURCE_RECORDINGS_H */
diff --git a/res/stasis_http/resource_sounds.c b/res/stasis_http/resource_sounds.c
new file mode 100644
index 000000000..a1808a159
--- /dev/null
+++ b/res/stasis_http/resource_sounds.c
@@ -0,0 +1,39 @@
+/*
+ * 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 /api-docs/sounds.{format} implementation- Sound resources
+ *
+ * \author David M. Lee, II <dlee@digium.com>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "resource_sounds.h"
+
+void stasis_http_get_sounds(struct ast_variable *headers, struct ast_get_sounds_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_sounds\n");
+}
+void stasis_http_get_stored_sound(struct ast_variable *headers, struct ast_get_stored_sound_args *args, struct stasis_http_response *response)
+{
+	ast_log(LOG_ERROR, "TODO: stasis_http_get_stored_sound\n");
+}
diff --git a/res/stasis_http/resource_sounds.h b/res/stasis_http/resource_sounds.h
new file mode 100644
index 000000000..f3010a920
--- /dev/null
+++ b/res/stasis_http/resource_sounds.h
@@ -0,0 +1,69 @@
+/*
+ * 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_http/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
+
+#include "asterisk/stasis_http.h"
+
+/*! \brief Argument struct for stasis_http_get_sounds() */
+struct ast_get_sounds_args {
+	const char *lang;
+	const char *format;
+};
+/*!
+ * \brief List all sounds.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_sounds(struct ast_variable *headers, struct ast_get_sounds_args *args, struct stasis_http_response *response);
+/*! \brief Argument struct for stasis_http_get_stored_sound() */
+struct ast_get_stored_sound_args {
+	/*! \brief Sound's id */
+	const char *sound_id;
+};
+/*!
+ * \brief Get a sound's details.
+ *
+ * \param headers HTTP headers
+ * \param args Swagger parameters
+ * \param[out] response HTTP response
+ */
+void stasis_http_get_stored_sound(struct ast_variable *headers, struct ast_get_stored_sound_args *args, struct stasis_http_response *response);
+
+#endif /* _ASTERISK_RESOURCE_SOUNDS_H */