Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
336 lines
10 KiB
C
336 lines
10 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2009 - 2014, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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* Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.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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* \author Joshua Colp <jcolp@digium.com>
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* \author Andreas 'MacBrody' Broadmann <andreas.brodmann@gmail.com>
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*
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* \brief RTP (Multicast and Unicast) Media Channel
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*
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* \ingroup channel_drivers
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_REGISTER_FILE()
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#include "asterisk/channel.h"
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#include "asterisk/module.h"
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#include "asterisk/pbx.h"
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#include "asterisk/acl.h"
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#include "asterisk/app.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/causes.h"
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#include "asterisk/format_cache.h"
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/* Forward declarations */
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static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
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static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
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static int rtp_call(struct ast_channel *ast, const char *dest, int timeout);
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static int rtp_hangup(struct ast_channel *ast);
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static struct ast_frame *rtp_read(struct ast_channel *ast);
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static int rtp_write(struct ast_channel *ast, struct ast_frame *f);
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/* Multicast channel driver declaration */
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static struct ast_channel_tech multicast_rtp_tech = {
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.type = "MulticastRTP",
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.description = "Multicast RTP Paging Channel Driver",
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.requester = multicast_rtp_request,
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.call = rtp_call,
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.hangup = rtp_hangup,
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.read = rtp_read,
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.write = rtp_write,
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};
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/* Unicast channel driver declaration */
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static struct ast_channel_tech unicast_rtp_tech = {
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.type = "UnicastRTP",
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.description = "Unicast RTP Media Channel Driver",
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.requester = unicast_rtp_request,
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.call = rtp_call,
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.hangup = rtp_hangup,
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.read = rtp_read,
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.write = rtp_write,
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};
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/*! \brief Function called when we should read a frame from the channel */
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static struct ast_frame *rtp_read(struct ast_channel *ast)
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{
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struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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int fdno = ast_channel_fdno(ast);
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switch (fdno) {
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case 0:
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return ast_rtp_instance_read(instance, 0);
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default:
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return &ast_null_frame;
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}
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}
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/*! \brief Function called when we should write a frame to the channel */
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static int rtp_write(struct ast_channel *ast, struct ast_frame *f)
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{
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struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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return ast_rtp_instance_write(instance, f);
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}
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/*! \brief Function called when we should actually call the destination */
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static int rtp_call(struct ast_channel *ast, const char *dest, int timeout)
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{
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struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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ast_queue_control(ast, AST_CONTROL_ANSWER);
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return ast_rtp_instance_activate(instance);
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}
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/*! \brief Function called when we should hang the channel up */
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static int rtp_hangup(struct ast_channel *ast)
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{
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struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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ast_rtp_instance_destroy(instance);
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ast_channel_tech_pvt_set(ast, NULL);
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return 0;
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}
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/*! \brief Function called when we should prepare to call the multicast destination */
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static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
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{
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char *parse;
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struct ast_rtp_instance *instance;
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struct ast_sockaddr control_address;
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struct ast_sockaddr destination_address;
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struct ast_channel *chan;
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struct ast_format_cap *caps = NULL;
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struct ast_format *fmt = NULL;
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(type);
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AST_APP_ARG(destination);
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AST_APP_ARG(control);
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);
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if (ast_strlen_zero(data)) {
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ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n");
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goto failure;
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}
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parse = ast_strdupa(data);
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AST_NONSTANDARD_APP_ARGS(args, parse, '/');
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fmt = ast_format_cap_get_format(cap, 0);
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ast_sockaddr_setnull(&control_address);
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if (!ast_strlen_zero(args.control) &&
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!ast_sockaddr_parse(&control_address, args.control, PARSE_PORT_REQUIRE)) {
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ast_log(LOG_ERROR, "Control address '%s' could not be parsed\n", args.control);
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goto failure;
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}
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if (!ast_sockaddr_parse(&destination_address, args.destination,
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PARSE_PORT_REQUIRE)) {
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ast_log(LOG_ERROR, "Destination address '%s' could not be parsed\n", args.destination);
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goto failure;
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}
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caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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if (!caps) {
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goto failure;
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}
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if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, args.type))) {
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ast_log(LOG_ERROR, "Could not create RTP instance for sending media to '%s'\n", args.destination);
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goto failure;
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}
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if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids, requestor, 0, "MulticastRTP/%p", instance))) {
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ast_rtp_instance_destroy(instance);
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goto failure;
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}
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ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
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ast_rtp_instance_set_remote_address(instance, &destination_address);
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ast_channel_tech_set(chan, &multicast_rtp_tech);
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ast_format_cap_append(caps, fmt, 0);
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ast_channel_nativeformats_set(chan, caps);
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ast_channel_set_writeformat(chan, fmt);
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ast_channel_set_rawwriteformat(chan, fmt);
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ast_channel_set_readformat(chan, fmt);
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ast_channel_set_rawreadformat(chan, fmt);
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ast_channel_tech_pvt_set(chan, instance);
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ast_channel_unlock(chan);
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ao2_ref(fmt, -1);
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ao2_ref(caps, -1);
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return chan;
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failure:
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ao2_cleanup(fmt);
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ao2_cleanup(caps);
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*cause = AST_CAUSE_FAILURE;
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return NULL;
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}
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/*! \brief Function called when we should prepare to call the unicast destination */
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static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
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{
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char *parse;
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struct ast_rtp_instance *instance;
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struct ast_sockaddr address;
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struct ast_sockaddr local_address;
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struct ast_channel *chan;
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struct ast_format_cap *caps = NULL;
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struct ast_format *fmt = NULL;
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(destination);
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AST_APP_ARG(engine);
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AST_APP_ARG(format);
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);
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if (ast_strlen_zero(data)) {
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goto failure;
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}
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parse = ast_strdupa(data);
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AST_NONSTANDARD_APP_ARGS(args, parse, '/');
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if (!ast_strlen_zero(args.format)) {
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fmt = ast_format_cache_get(args.format);
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} else {
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fmt = ast_format_cap_get_format(cap, 0);
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}
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if (!fmt) {
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ast_log(LOG_ERROR, "No format specified for sending RTP to '%s'\n", args.destination);
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goto failure;
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}
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if (!ast_sockaddr_parse(&address, args.destination,
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PARSE_PORT_REQUIRE)) {
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ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
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goto failure;
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}
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caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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if (!caps) {
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goto failure;
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}
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ast_ouraddrfor(&address, &local_address);
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if (!(instance = ast_rtp_instance_new(args.engine, NULL, &local_address, NULL))) {
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ast_log(LOG_ERROR, "Could not create RTP instance for sending media to '%s'\n", args.destination);
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goto failure;
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}
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if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids, requestor, 0, "UnicastRTP/%s-%p", args.destination, instance))) {
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ast_rtp_instance_destroy(instance);
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goto failure;
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}
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ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
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ast_rtp_instance_set_remote_address(instance, &address);
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ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(instance, 0));
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ast_channel_tech_set(chan, &unicast_rtp_tech);
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ast_format_cap_append(caps, fmt, 0);
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ast_channel_nativeformats_set(chan, caps);
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ast_channel_set_writeformat(chan, fmt);
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ast_channel_set_rawwriteformat(chan, fmt);
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ast_channel_set_readformat(chan, fmt);
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ast_channel_set_rawreadformat(chan, fmt);
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ast_channel_tech_pvt_set(chan, instance);
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pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS", ast_sockaddr_stringify_addr(&local_address));
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ast_rtp_instance_get_local_address(instance, &local_address);
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pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT", ast_sockaddr_stringify_port(&local_address));
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ast_channel_unlock(chan);
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ao2_ref(fmt, -1);
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ao2_ref(caps, -1);
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return chan;
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failure:
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ao2_cleanup(fmt);
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ao2_cleanup(caps);
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*cause = AST_CAUSE_FAILURE;
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return NULL;
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}
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/*! \brief Function called when our module is unloaded */
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static int unload_module(void)
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{
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ast_channel_unregister(&multicast_rtp_tech);
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ao2_cleanup(multicast_rtp_tech.capabilities);
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multicast_rtp_tech.capabilities = NULL;
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ast_channel_unregister(&unicast_rtp_tech);
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ao2_cleanup(unicast_rtp_tech.capabilities);
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unicast_rtp_tech.capabilities = NULL;
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return 0;
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}
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/*! \brief Function called when our module is loaded */
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static int load_module(void)
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{
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if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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return AST_MODULE_LOAD_DECLINE;
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}
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ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
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if (ast_channel_register(&multicast_rtp_tech)) {
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ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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if (!(unicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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ast_format_cap_append_by_type(unicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
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if (ast_channel_register(&unicast_rtp_tech)) {
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ast_log(LOG_ERROR, "Unable to register channel class 'UnicastRTP'\n");
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTP Media Channel",
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.support_level = AST_MODULE_SUPPORT_CORE,
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_CHANNEL_DRIVER,
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);
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