........ r398638 | dlee | 2013-09-09 14:01:54 -0500 (Mon, 09 Sep 2013) | 1 line Added note about expected behavior of originate ........ r398639 | dlee | 2013-09-09 14:02:27 -0500 (Mon, 09 Sep 2013) | 1 line Added note about expected behavior of originate (the rest of the commit) ........ Merged revisions 398638-398639 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
333 lines
12 KiB
C
333 lines
12 KiB
C
/*
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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/ari/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/ari_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/ari.h"
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/*! \brief Argument struct for ast_ari_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 ast_ari_get_channels(struct ast_variable *headers, struct ast_get_channels_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_originate() */
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struct ast_originate_args {
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/*! \brief Endpoint to call. */
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const char *endpoint;
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/*! \brief The extension to dial after the endpoint answers */
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const char *extension;
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/*! \brief The context to dial after the endpoint answers. If omitted, uses 'default' */
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const char *context;
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/*! \brief The priority to dial after the endpoint answers. If omitted, uses 1 */
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long priority;
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/*! \brief The application name to pass to the Stasis application. */
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const char *app;
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/*! \brief The application arguments to pass to the Stasis application. */
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const char *app_args;
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/*! \brief CallerID to use when dialing the endpoint or extension. */
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const char *caller_id;
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/*! \brief Timeout (in seconds) before giving up dialing, or -1 for no timeout. */
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int timeout;
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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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* The new channel is not created until the dialed party picks up. Not wanting to block this request indefinitely, this request returns immediately with a 204 No Content. When the channel is created, a StasisStart event is sent with the provided app and appArgs. In the event of a failure (timeout, busy, etc.), an OriginationFailed event is sent.
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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 ast_ari_originate(struct ast_variable *headers, struct ast_originate_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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 ast_ari_get_channel(struct ast_variable *headers, struct ast_get_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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 ast_ari_delete_channel(struct ast_variable *headers, struct ast_delete_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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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/*! \brief Timeout (in seconds) before giving up dialing, or -1 for no timeout. */
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int timeout;
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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 ast_ari_dial(struct ast_variable *headers, struct ast_dial_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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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/*! \brief The context to continue to. */
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const char *context;
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/*! \brief The extension to continue to. */
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const char *extension;
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/*! \brief The priority to continue to. */
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int priority;
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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 ast_ari_continue_in_dialplan(struct ast_variable *headers, struct ast_continue_in_dialplan_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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 ast_ari_answer_channel(struct ast_variable *headers, struct ast_answer_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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 ast_ari_mute_channel(struct ast_variable *headers, struct ast_mute_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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 ast_ari_unmute_channel(struct ast_variable *headers, struct ast_unmute_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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 ast_ari_hold_channel(struct ast_variable *headers, struct ast_hold_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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 ast_ari_unhold_channel(struct ast_variable *headers, struct ast_unhold_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_moh_start_channel() */
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struct ast_moh_start_channel_args {
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/*! \brief Channel's id */
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const char *channel_id;
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/*! \brief Music on hold class to use */
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const char *moh_class;
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};
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/*!
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* \brief Play music on hold to a channel.
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*
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* Using media operations such as playOnChannel on a channel playing MOH in this manner will suspend MOH without resuming automatically. If continuing music on hold is desired, the stasis application must reinitiate music on 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 ast_ari_moh_start_channel(struct ast_variable *headers, struct ast_moh_start_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_moh_stop_channel() */
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struct ast_moh_stop_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 Stop playing music on hold to 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 ast_ari_moh_stop_channel(struct ast_variable *headers, struct ast_moh_stop_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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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/*! \brief For sounds, selects language for sound. */
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const char *lang;
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/*! \brief Number of media to skip before playing. */
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int offsetms;
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/*! \brief Number of milliseconds to skip for forward/reverse operations. */
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int skipms;
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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 ast_ari_play_on_channel(struct ast_variable *headers, struct ast_play_on_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_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 Action to take if a recording with the same name already exists. */
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const char *if_exists;
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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 ast_ari_record_channel(struct ast_variable *headers, struct ast_record_channel_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_get_channel_var() */
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struct ast_get_channel_var_args {
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/*! \brief Channel's id */
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const char *channel_id;
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/*! \brief The channel variable or function to get */
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const char *variable;
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};
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/*!
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* \brief Get the value of a channel variable or function.
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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 ast_ari_get_channel_var(struct ast_variable *headers, struct ast_get_channel_var_args *args, struct ast_ari_response *response);
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/*! \brief Argument struct for ast_ari_set_channel_var() */
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struct ast_set_channel_var_args {
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/*! \brief Channel's id */
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const char *channel_id;
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/*! \brief The channel variable or function to set */
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const char *variable;
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/*! \brief The value to set the variable to */
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const char *value;
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};
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/*!
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* \brief Set the value of a channel variable or function.
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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 ast_ari_set_channel_var(struct ast_variable *headers, struct ast_set_channel_var_args *args, struct ast_ari_response *response);
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#endif /* _ASTERISK_RESOURCE_CHANNELS_H */
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