735b30ad71
The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*" * All files that were "res_sip*" are now "res_pjsip*" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
129 lines
3.4 KiB
C
129 lines
3.4 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2013, malleable, llc.
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*
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* Sean Bright <sean@malleable.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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/*** MODULEINFO
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<depend>pjproject</depend>
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<depend>res_pjsip</depend>
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<depend>res_pjsip_session</depend>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <pjsip.h>
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#include <pjsip_ua.h>
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#include "asterisk/features.h"
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#include "asterisk/res_pjsip.h"
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#include "asterisk/res_pjsip_session.h"
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#include "asterisk/module.h"
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#include "asterisk/features_config.h"
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static void send_response(struct ast_sip_session *session, int code, struct pjsip_rx_data *rdata)
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{
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pjsip_tx_data *tdata;
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if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, code, NULL, &tdata) == PJ_SUCCESS) {
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struct pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
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pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
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}
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}
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static int handle_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
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{
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static const pj_str_t rec_str = { "Record", 6 };
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pjsip_generic_string_hdr *record;
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int feature_res;
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char feature_code[AST_FEATURE_MAX_LEN];
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const char *feature;
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char *digit;
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record = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &rec_str, NULL);
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/* If we don't have Record header, we have nothing to do */
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if (!record) {
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return 0;
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}
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if (!pj_stricmp2(&record->hvalue, "on")) {
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feature = session->endpoint->info.recording.onfeature;
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} else if (!pj_stricmp2(&record->hvalue, "off")) {
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feature = session->endpoint->info.recording.offfeature;
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} else {
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/* Don't send response because another module may handle this */
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return 0;
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}
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if (!session->channel) {
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send_response(session, 481, rdata);
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return 0;
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}
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/* Is this endpoint configured with One Touch Recording? */
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if (!session->endpoint->info.recording.enabled || ast_strlen_zero(feature)) {
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send_response(session, 403, rdata);
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return 0;
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}
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ast_channel_lock(session->channel);
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feature_res = ast_get_feature(session->channel, feature, feature_code, sizeof(feature_code));
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ast_channel_unlock(session->channel);
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if (feature_res || ast_strlen_zero(feature_code)) {
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send_response(session, 403, rdata);
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return 0;
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}
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for (digit = feature_code; *digit; ++digit) {
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struct ast_frame f = { AST_FRAME_DTMF, .subclass.integer = *digit, .len = 100 };
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ast_queue_frame(session->channel, &f);
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}
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send_response(session, 200, rdata);
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return 0;
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}
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static struct ast_sip_session_supplement info_supplement = {
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.method = "INFO",
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.incoming_request = handle_incoming_request,
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};
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static int load_module(void)
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{
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if (ast_sip_session_register_supplement(&info_supplement)) {
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ast_log(LOG_ERROR, "Unable to register One Touch Recording supplement\n");
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return AST_MODULE_LOAD_FAILURE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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static int unload_module(void)
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{
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ast_sip_session_unregister_supplement(&info_supplement);
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return 0;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP INFO One Touch Recording Support",
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.load = load_module,
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.unload = unload_module,
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.load_pri = AST_MODPRI_APP_DEPEND,
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);
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