asterisk/res/res_pjsip_dlg_options.c
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro
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
2015-04-13 03:48:57 -04:00

108 lines
3.1 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2015, Digium, Inc.
*
* Yaron Nahum <nachum.yaron@gmail.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.
*/
/*** MODULEINFO
<depend>pjproject</depend>
<depend>res_pjsip</depend>
<depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_REGISTER_FILE()
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjlib.h>
#include "asterisk/module.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#define DEFAULT_LANGUAGE "en"
#define DEFAULT_ENCODING "text/plain"
static int options_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
pjsip_tx_data *tdata;
pj_status_t status;
const pjsip_hdr *hdr;
pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
status = pjsip_dlg_create_response(session->inv_session->dlg, rdata, 200, NULL,&tdata);
if (status != PJ_SUCCESS) {
ast_log(LOG_ERROR, "Unable to create response (%d)\n", status);
return status;
}
/* Add appropriate headers */
if ((hdr = pjsip_endpt_get_capability(endpt, PJSIP_H_ACCEPT, NULL))) {
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)pjsip_hdr_clone(tdata->pool, hdr));
}
if ((hdr = pjsip_endpt_get_capability(endpt, PJSIP_H_ALLOW, NULL))) {
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)pjsip_hdr_clone(tdata->pool, hdr));
}
if ((hdr = pjsip_endpt_get_capability(endpt, PJSIP_H_SUPPORTED, NULL))) {
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)pjsip_hdr_clone(tdata->pool, hdr));
}
/*
* XXX TODO: pjsip doesn't care a lot about either of these headers -
* while it provides specific methods to create them, they are defined
* to be the standard string header creation. We never did add them
* in chan_sip, although RFC 3261 says they SHOULD. Hard coded here.
*/
ast_sip_add_header(tdata, "Accept-Encoding", DEFAULT_ENCODING);
ast_sip_add_header(tdata, "Accept-Language", DEFAULT_LANGUAGE);
status = pjsip_dlg_send_response(session->inv_session->dlg, pjsip_rdata_get_tsx(rdata), tdata);
if (status != PJ_SUCCESS) {
ast_log(LOG_ERROR, "Unable to send response (%d)\n", status);
}
return status;
}
static struct ast_sip_session_supplement dlg_options_supplement = {
.method = "OPTIONS",
.incoming_request = options_incoming_request,
};
static int load_module(void)
{
CHECK_PJSIP_MODULE_LOADED();
if (ast_sip_session_register_supplement(&dlg_options_supplement)) {
return AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_sip_session_unregister_supplement(&dlg_options_supplement);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP OPTIONS in dialog handler",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
);