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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
This commit is contained in:
David M. Lee
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res/stasis_http/resource_channels.h
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244
res/stasis_http/resource_channels.h
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2012 - 2013, Digium, Inc.
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*
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* David M. Lee, II <dlee@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Generated file - declares stubs to be implemented in
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* res/stasis_http/resource_channels.c
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*
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* Channel resources
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*
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* \author David M. Lee, II <dlee@digium.com>
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*/
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/*
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* !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
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* !!!!! DO NOT EDIT !!!!!
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* !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
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* This file is generated by a mustache template. Please see the original
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* template in rest-api-templates/stasis_http_resource.h.mustache
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*/
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#ifndef _ASTERISK_RESOURCE_CHANNELS_H
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#define _ASTERISK_RESOURCE_CHANNELS_H
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#include "asterisk/stasis_http.h"
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/*! \brief Argument struct for stasis_http_get_channels() */
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struct ast_get_channels_args {
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};
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/*!
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* \brief List active channels.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_get_channels(struct ast_variable *headers, struct ast_get_channels_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_originate() */
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struct ast_originate_args {
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/*! \brief Endpoint to call. If not specified, originate is routed via dialplan */
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const char *endpoint;
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/*! \brief Extension to dial */
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const char *extension;
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/*! \brief When routing via dialplan, the context use. If omitted, uses 'default' */
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const char *context;
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};
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/*!
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* \brief Create a new channel (originate).
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_originate(struct ast_variable *headers, struct ast_originate_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_get_channel() */
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struct ast_get_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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};
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/*!
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* \brief Channel details.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_get_channel(struct ast_variable *headers, struct ast_get_channel_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_delete_channel() */
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struct ast_delete_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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};
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/*!
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* \brief Delete (i.e. hangup) a channel.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_delete_channel(struct ast_variable *headers, struct ast_delete_channel_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_dial() */
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struct ast_dial_args {
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/*! \brief Channel's id */
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const char *channel_id;
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/*! \brief Endpoint to call. If not specified, dial is routed via dialplan */
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const char *endpoint;
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/*! \brief Extension to dial */
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const char *extension;
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/*! \brief When routing via dialplan, the context use. If omitted, uses 'default' */
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const char *context;
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};
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/*!
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* \brief Create a new channel (originate) and bridge to this channel.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_dial(struct ast_variable *headers, struct ast_dial_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_continue_in_dialplan() */
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struct ast_continue_in_dialplan_args {
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/*! \brief Channel's id */
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const char *channel_id;
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};
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/*!
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* \brief Exit application; continue execution in the dialplan.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_continue_in_dialplan(struct ast_variable *headers, struct ast_continue_in_dialplan_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_answer_channel() */
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struct ast_answer_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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};
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/*!
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* \brief Answer a channel.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_answer_channel(struct ast_variable *headers, struct ast_answer_channel_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_mute_channel() */
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struct ast_mute_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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/*! \brief Direction in which to mute audio */
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const char *direction;
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};
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/*!
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* \brief Mute a channel.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_mute_channel(struct ast_variable *headers, struct ast_mute_channel_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_unmute_channel() */
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struct ast_unmute_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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/*! \brief Direction in which to unmute audio */
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const char *direction;
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};
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/*!
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* \brief Unmute a channel.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_unmute_channel(struct ast_variable *headers, struct ast_unmute_channel_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_hold_channel() */
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struct ast_hold_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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};
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/*!
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* \brief Hold a channel.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_hold_channel(struct ast_variable *headers, struct ast_hold_channel_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_unhold_channel() */
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struct ast_unhold_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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};
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/*!
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* \brief Remove a channel from hold.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_unhold_channel(struct ast_variable *headers, struct ast_unhold_channel_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_play_on_channel() */
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struct ast_play_on_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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/*! \brief Media's URI to play. */
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const char *media;
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};
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/*!
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* \brief Start playback of media.
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*
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* 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.)
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_play_on_channel(struct ast_variable *headers, struct ast_play_on_channel_args *args, struct stasis_http_response *response);
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/*! \brief Argument struct for stasis_http_record_channel() */
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struct ast_record_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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/*! \brief Recording's filename */
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const char *name;
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/*! \brief Format to encode audio in */
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const char *format;
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/*! \brief Maximum duration of the recording, in seconds. 0 for no limit */
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int max_duration_seconds;
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/*! \brief Maximum duration of silence, in seconds. 0 for no limit */
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int max_silence_seconds;
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/*! \brief If true, and recording already exists, append to recording */
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int append;
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/*! \brief Play beep when recording begins */
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int beep;
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/*! \brief DTMF input to terminate recording */
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const char *terminate_on;
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};
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/*!
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* \brief Start a recording.
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*
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* 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.
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*
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* \param headers HTTP headers
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* \param args Swagger parameters
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* \param[out] response HTTP response
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*/
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void stasis_http_record_channel(struct ast_variable *headers, struct ast_record_channel_args *args, struct stasis_http_response *response);
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#endif /* _ASTERISK_RESOURCE_CHANNELS_H */
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