3d63833bd6
Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1472 lines
41 KiB
C
1472 lines
41 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2013, Digium, Inc.
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*
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* Joshua Colp <jcolp@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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* \author Joshua Colp <jcolp@digium.com>
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*
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* \brief Gulp SIP Channel Driver
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*
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* \ingroup channel_drivers
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*/
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/*** MODULEINFO
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<depend>pjproject</depend>
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<depend>res_sip</depend>
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<depend>res_sip_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 <pjlib.h>
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/lock.h"
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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/rtp_engine.h"
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#include "asterisk/acl.h"
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#include "asterisk/callerid.h"
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#include "asterisk/file.h"
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#include "asterisk/cli.h"
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#include "asterisk/app.h"
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#include "asterisk/musiconhold.h"
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#include "asterisk/causes.h"
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#include "asterisk/taskprocessor.h"
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#include "asterisk/res_sip.h"
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#include "asterisk/res_sip_session.h"
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/*** DOCUMENTATION
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<function name="GULP_DIAL_CONTACTS" language="en_US">
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<synopsis>
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Return a dial string for dialing all contacts on an AOR.
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</synopsis>
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<syntax>
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<parameter name="endpoint" required="true">
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<para>Name of the endpoint</para>
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</parameter>
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<parameter name="aor" required="false">
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<para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
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</parameter>
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<parameter name="request_user" required="false">
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<para>Optional request user to use in the request URI</para>
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</parameter>
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</syntax>
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<description>
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<para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
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</description>
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</function>
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***/
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static const char desc[] = "Gulp SIP Channel";
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static const char channel_type[] = "Gulp";
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/*!
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* \brief Positions of various media
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*/
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enum sip_session_media_position {
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/*! \brief First is audio */
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SIP_MEDIA_AUDIO = 0,
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/*! \brief Second is video */
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SIP_MEDIA_VIDEO,
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/*! \brief Last is the size for media details */
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SIP_MEDIA_SIZE,
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};
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struct gulp_pvt {
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struct ast_sip_session *session;
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struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
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};
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static void gulp_pvt_dtor(void *obj)
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{
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struct gulp_pvt *pvt = obj;
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int i;
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ao2_cleanup(pvt->session);
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pvt->session = NULL;
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for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
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ao2_cleanup(pvt->media[i]);
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pvt->media[i] = NULL;
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}
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}
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/* \brief Asterisk core interaction functions */
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static struct ast_channel *gulp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
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static int gulp_sendtext(struct ast_channel *ast, const char *text);
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static int gulp_digit_begin(struct ast_channel *ast, char digit);
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static int gulp_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
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static int gulp_call(struct ast_channel *ast, const char *dest, int timeout);
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static int gulp_hangup(struct ast_channel *ast);
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static int gulp_answer(struct ast_channel *ast);
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static struct ast_frame *gulp_read(struct ast_channel *ast);
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static int gulp_write(struct ast_channel *ast, struct ast_frame *f);
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static int gulp_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
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static int gulp_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
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/*! \brief PBX interface structure for channel registration */
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static struct ast_channel_tech gulp_tech = {
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.type = channel_type,
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.description = "Gulp SIP Channel Driver",
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.requester = gulp_request,
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.send_text = gulp_sendtext,
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.send_digit_begin = gulp_digit_begin,
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.send_digit_end = gulp_digit_end,
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.call = gulp_call,
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.hangup = gulp_hangup,
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.answer = gulp_answer,
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.read = gulp_read,
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.write = gulp_write,
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.write_video = gulp_write,
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.exception = gulp_read,
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.indicate = gulp_indicate,
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.fixup = gulp_fixup,
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.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
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};
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/*! \brief SIP session interaction functions */
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static void gulp_session_begin(struct ast_sip_session *session);
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static void gulp_session_end(struct ast_sip_session *session);
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static int gulp_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
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static void gulp_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
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/*! \brief SIP session supplement structure */
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static struct ast_sip_session_supplement gulp_supplement = {
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.method = "INVITE",
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.priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
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.session_begin = gulp_session_begin,
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.session_end = gulp_session_end,
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.incoming_request = gulp_incoming_request,
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.incoming_response = gulp_incoming_response,
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};
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static int gulp_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
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static struct ast_sip_session_supplement gulp_ack_supplement = {
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.method = "ACK",
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.priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
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.incoming_request = gulp_incoming_ack,
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};
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/*! \brief Dialplan function for constructing a dial string for calling all contacts */
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static int gulp_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
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{
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RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
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RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
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const char *aor_name;
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char *rest;
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(endpoint_name);
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AST_APP_ARG(aor_name);
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AST_APP_ARG(request_user);
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);
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AST_STANDARD_APP_ARGS(args, data);
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if (ast_strlen_zero(args.endpoint_name)) {
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ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
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return -1;
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} else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
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ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
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return -1;
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}
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aor_name = S_OR(args.aor_name, endpoint->aors);
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if (ast_strlen_zero(aor_name)) {
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ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
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return -1;
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} else if (!(dial = ast_str_create(len))) {
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ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
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return -1;
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} else if (!(rest = ast_strdupa(aor_name))) {
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ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
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return -1;
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}
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while ((aor_name = strsep(&rest, ","))) {
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RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
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RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
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struct ao2_iterator it_contacts;
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struct ast_sip_contact *contact;
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if (!aor) {
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/* If the AOR provided is not found skip it, there may be more */
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continue;
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} else if (!(contacts = ast_sip_location_retrieve_aor_contacts(aor))) {
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/* No contacts are available, skip it as well */
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continue;
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} else if (!ao2_container_count(contacts)) {
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/* We were given a container but no contacts are in it... */
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continue;
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}
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it_contacts = ao2_iterator_init(contacts, 0);
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for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
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ast_str_append(&dial, -1, "Gulp/");
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if (!ast_strlen_zero(args.request_user)) {
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ast_str_append(&dial, -1, "%s@", args.request_user);
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}
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ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
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}
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ao2_iterator_destroy(&it_contacts);
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}
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/* Trim the '&' at the end off */
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ast_str_truncate(dial, ast_str_strlen(dial) - 1);
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ast_copy_string(buf, ast_str_buffer(dial), len);
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return 0;
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}
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static struct ast_custom_function gulp_dial_contacts_function = {
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.name = "GULP_DIAL_CONTACTS",
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.read = gulp_dial_contacts,
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};
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/*! \brief Function called by RTP engine to get local audio RTP peer */
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static enum ast_rtp_glue_result gulp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
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{
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struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
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struct ast_sip_endpoint *endpoint;
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if (!pvt || !pvt->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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endpoint = pvt->session->endpoint;
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*instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
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ao2_ref(*instance, +1);
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ast_assert(endpoint != NULL);
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if (endpoint->direct_media) {
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return AST_RTP_GLUE_RESULT_REMOTE;
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}
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return AST_RTP_GLUE_RESULT_LOCAL;
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}
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/*! \brief Function called by RTP engine to get local video RTP peer */
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static enum ast_rtp_glue_result gulp_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
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{
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struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
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if (!pvt || !pvt->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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*instance = pvt->media[SIP_MEDIA_VIDEO]->rtp;
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ao2_ref(*instance, +1);
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return AST_RTP_GLUE_RESULT_LOCAL;
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}
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/*! \brief Function called by RTP engine to get peer capabilities */
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static void gulp_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
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{
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struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
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ast_format_cap_copy(result, pvt->session->endpoint->codecs);
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}
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static int send_direct_media_request(void *data)
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{
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RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
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return ast_sip_session_refresh(session, NULL, NULL, session->endpoint->direct_media_method, 1);
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}
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static struct ast_datastore_info direct_media_mitigation_info = { };
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static int direct_media_mitigate_glare(struct ast_sip_session *session)
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{
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RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
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if (session->endpoint->direct_media_glare_mitigation ==
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AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
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return 0;
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}
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datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
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if (!datastore) {
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return 0;
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}
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/* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
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ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
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if ((session->endpoint->direct_media_glare_mitigation ==
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AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING &&
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session->inv_session->role == PJSIP_ROLE_UAC) ||
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(session->endpoint->direct_media_glare_mitigation ==
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AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING &&
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session->inv_session->role == PJSIP_ROLE_UAS)) {
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return 1;
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}
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return 0;
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}
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static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp,
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struct ast_sip_session_media *media, int rtcp_fd)
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{
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int changed = 0;
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if (rtp) {
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changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
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if (media->rtp) {
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ast_channel_set_fd(chan, rtcp_fd, -1);
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ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
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}
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} else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
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ast_sockaddr_setnull(&media->direct_media_addr);
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changed = 1;
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if (media->rtp) {
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ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
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ast_channel_set_fd(chan, rtcp_fd, ast_rtp_instance_fd(media->rtp, 1));
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}
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}
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return changed;
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}
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/*! \brief Function called by RTP engine to change where the remote party should send media */
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static int gulp_set_rtp_peer(struct ast_channel *chan,
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struct ast_rtp_instance *rtp,
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struct ast_rtp_instance *vrtp,
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struct ast_rtp_instance *tpeer,
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const struct ast_format_cap *cap,
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int nat_active)
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{
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struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
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struct ast_sip_session *session = pvt->session;
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int changed = 0;
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/* Don't try to do any direct media shenanigans on early bridges */
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if ((rtp || vrtp || tpeer) && !ast_bridged_channel(chan)) {
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return 0;
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}
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if (nat_active && session->endpoint->disable_direct_media_on_nat) {
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return 0;
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}
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if (pvt->media[SIP_MEDIA_AUDIO]) {
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changed |= check_for_rtp_changes(chan, rtp, pvt->media[SIP_MEDIA_AUDIO], 1);
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}
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if (pvt->media[SIP_MEDIA_VIDEO]) {
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changed |= check_for_rtp_changes(chan, vrtp, pvt->media[SIP_MEDIA_VIDEO], 3);
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}
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if (direct_media_mitigate_glare(session)) {
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return 0;
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}
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if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(session->direct_media_cap, cap)) {
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ast_format_cap_copy(session->direct_media_cap, cap);
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changed = 1;
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}
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if (changed) {
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ao2_ref(session, +1);
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ast_sip_push_task(session->serializer, send_direct_media_request, session);
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}
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return 0;
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}
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/*! \brief Local glue for interacting with the RTP engine core */
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static struct ast_rtp_glue gulp_rtp_glue = {
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.type = "Gulp",
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.get_rtp_info = gulp_get_rtp_peer,
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.get_vrtp_info = gulp_get_vrtp_peer,
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.get_codec = gulp_get_codec,
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.update_peer = gulp_set_rtp_peer,
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};
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|
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/*! \brief Function called to create a new Gulp Asterisk channel */
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static struct ast_channel *gulp_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const char *linkedid, const char *cid_name)
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{
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struct ast_channel *chan;
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struct ast_format fmt;
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struct gulp_pvt *pvt;
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if (!(pvt = ao2_alloc(sizeof(*pvt), gulp_pvt_dtor))) {
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return NULL;
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}
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if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "Gulp/%s-%.*s", ast_sorcery_object_get_id(session->endpoint),
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(int)session->inv_session->dlg->call_id->id.slen, session->inv_session->dlg->call_id->id.ptr))) {
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ao2_cleanup(pvt);
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return NULL;
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}
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ast_channel_tech_set(chan, &gulp_tech);
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ao2_ref(session, +1);
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pvt->session = session;
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/* If res_sip_session is ever updated to create/destroy ast_sip_session_media
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* during a call such as if multiple same-type stream support is introduced,
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* these will need to be recaptured as well */
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pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
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pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
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ast_channel_tech_pvt_set(chan, pvt);
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if (ast_format_cap_is_empty(session->req_caps) || !ast_format_cap_has_joint(session->req_caps, session->endpoint->codecs)) {
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ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->codecs);
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} else {
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ast_format_cap_copy(ast_channel_nativeformats(chan), session->req_caps);
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}
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|
|
|
ast_codec_choose(&session->endpoint->prefs, ast_channel_nativeformats(chan), 1, &fmt);
|
|
ast_format_copy(ast_channel_writeformat(chan), &fmt);
|
|
ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
|
|
ast_format_copy(ast_channel_readformat(chan), &fmt);
|
|
ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
|
|
|
|
if (state == AST_STATE_RING) {
|
|
ast_channel_rings_set(chan, 1);
|
|
}
|
|
|
|
ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
|
|
|
|
ast_channel_context_set(chan, session->endpoint->context);
|
|
ast_channel_exten_set(chan, S_OR(exten, "s"));
|
|
ast_channel_priority_set(chan, 1);
|
|
|
|
return chan;
|
|
}
|
|
|
|
static int answer(void *data)
|
|
{
|
|
pj_status_t status;
|
|
pjsip_tx_data *packet;
|
|
struct ast_sip_session *session = data;
|
|
|
|
if ((status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet)) == PJ_SUCCESS) {
|
|
ast_sip_session_send_response(session, packet);
|
|
}
|
|
|
|
ao2_ref(session, -1);
|
|
|
|
return (status == PJ_SUCCESS) ? 0 : -1;
|
|
}
|
|
|
|
/*! \brief Function called by core when we should answer a Gulp session */
|
|
static int gulp_answer(struct ast_channel *ast)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session = pvt->session;
|
|
|
|
if (ast_channel_state(ast) == AST_STATE_UP) {
|
|
return 0;
|
|
}
|
|
|
|
ast_setstate(ast, AST_STATE_UP);
|
|
|
|
ao2_ref(session, +1);
|
|
if (ast_sip_push_task(session->serializer, answer, session)) {
|
|
ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
|
|
ao2_cleanup(session);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to read any waiting frames */
|
|
static struct ast_frame *gulp_read(struct ast_channel *ast)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
|
|
struct ast_frame *f;
|
|
struct ast_sip_session_media *media = NULL;
|
|
int rtcp = 0;
|
|
int fdno = ast_channel_fdno(ast);
|
|
|
|
switch (fdno) {
|
|
case 0:
|
|
media = pvt->media[SIP_MEDIA_AUDIO];
|
|
break;
|
|
case 1:
|
|
media = pvt->media[SIP_MEDIA_AUDIO];
|
|
rtcp = 1;
|
|
break;
|
|
case 2:
|
|
media = pvt->media[SIP_MEDIA_VIDEO];
|
|
break;
|
|
case 3:
|
|
media = pvt->media[SIP_MEDIA_VIDEO];
|
|
rtcp = 1;
|
|
break;
|
|
}
|
|
|
|
if (!media || !media->rtp) {
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
f = ast_rtp_instance_read(media->rtp, rtcp);
|
|
|
|
if (f && f->frametype == AST_FRAME_VOICE) {
|
|
if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &f->subclass.format))) {
|
|
ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
|
|
ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
|
|
ast_set_read_format(ast, ast_channel_readformat(ast));
|
|
ast_set_write_format(ast, ast_channel_writeformat(ast));
|
|
}
|
|
}
|
|
|
|
return f;
|
|
}
|
|
|
|
/*! \brief Function called by core to write frames */
|
|
static int gulp_write(struct ast_channel *ast, struct ast_frame *frame)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session_media *media;
|
|
int res = 0;
|
|
|
|
switch (frame->frametype) {
|
|
case AST_FRAME_VOICE:
|
|
media = pvt->media[SIP_MEDIA_AUDIO];
|
|
|
|
if (!media) {
|
|
return 0;
|
|
}
|
|
if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
|
|
char buf[256];
|
|
|
|
ast_log(LOG_WARNING,
|
|
"Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
|
|
ast_getformatname(&frame->subclass.format),
|
|
ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
|
|
ast_getformatname(ast_channel_readformat(ast)),
|
|
ast_getformatname(ast_channel_writeformat(ast)));
|
|
return 0;
|
|
}
|
|
if (media->rtp) {
|
|
res = ast_rtp_instance_write(media->rtp, frame);
|
|
}
|
|
break;
|
|
case AST_FRAME_VIDEO:
|
|
if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
|
|
res = ast_rtp_instance_write(media->rtp, frame);
|
|
}
|
|
break;
|
|
default:
|
|
ast_log(LOG_WARNING, "Can't send %d type frames with Gulp\n", frame->frametype);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
struct fixup_data {
|
|
struct ast_sip_session *session;
|
|
struct ast_channel *chan;
|
|
};
|
|
|
|
static int fixup(void *data)
|
|
{
|
|
struct fixup_data *fix_data = data;
|
|
|
|
fix_data->session->channel = fix_data->chan;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to change the underlying owner channel */
|
|
static int gulp_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(newchan);
|
|
struct ast_sip_session *session = pvt->session;
|
|
struct fixup_data fix_data;
|
|
|
|
fix_data.session = session;
|
|
fix_data.chan = newchan;
|
|
|
|
if (session->channel != oldchan) {
|
|
return -1;
|
|
}
|
|
|
|
if (ast_sip_push_task_synchronous(session->serializer, fixup, &fix_data)) {
|
|
ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
struct indicate_data {
|
|
struct ast_sip_session *session;
|
|
int condition;
|
|
int response_code;
|
|
void *frame_data;
|
|
size_t datalen;
|
|
};
|
|
|
|
static void indicate_data_destroy(void *obj)
|
|
{
|
|
struct indicate_data *ind_data = obj;
|
|
|
|
ast_free(ind_data->frame_data);
|
|
ao2_ref(ind_data->session, -1);
|
|
}
|
|
|
|
static struct indicate_data *indicate_data_alloc(struct ast_sip_session *session,
|
|
int condition, int response_code, const void *frame_data, size_t datalen)
|
|
{
|
|
struct indicate_data *ind_data = ao2_alloc(sizeof(*ind_data), indicate_data_destroy);
|
|
|
|
if (!ind_data) {
|
|
return NULL;
|
|
}
|
|
|
|
ind_data->frame_data = ast_malloc(datalen);
|
|
if (!ind_data->frame_data) {
|
|
ao2_ref(ind_data, -1);
|
|
return NULL;
|
|
}
|
|
|
|
memcpy(ind_data->frame_data, frame_data, datalen);
|
|
ind_data->datalen = datalen;
|
|
ind_data->condition = condition;
|
|
ind_data->response_code = response_code;
|
|
ao2_ref(session, +1);
|
|
ind_data->session = session;
|
|
|
|
return ind_data;
|
|
}
|
|
|
|
static int indicate(void *data)
|
|
{
|
|
pjsip_tx_data *packet = NULL;
|
|
struct indicate_data *ind_data = data;
|
|
struct ast_sip_session *session = ind_data->session;
|
|
int response_code = ind_data->response_code;
|
|
|
|
if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
|
|
ast_sip_session_send_response(session, packet);
|
|
}
|
|
|
|
ao2_ref(ind_data, -1);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Send SIP INFO with video update request */
|
|
static int transmit_info_with_vidupdate(void *data)
|
|
{
|
|
const char * xml =
|
|
"<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
|
|
" <media_control>\r\n"
|
|
" <vc_primitive>\r\n"
|
|
" <to_encoder>\r\n"
|
|
" <picture_fast_update/>\r\n"
|
|
" </to_encoder>\r\n"
|
|
" </vc_primitive>\r\n"
|
|
" </media_control>\r\n";
|
|
|
|
const struct ast_sip_body body = {
|
|
.type = "application",
|
|
.subtype = "media_control+xml",
|
|
.body_text = xml
|
|
};
|
|
|
|
struct ast_sip_session *session = data;
|
|
struct pjsip_tx_data *tdata;
|
|
|
|
if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
|
|
ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
|
|
return -1;
|
|
}
|
|
if (ast_sip_add_body(tdata, &body)) {
|
|
ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
|
|
return -1;
|
|
}
|
|
ast_sip_session_send_request(session, tdata);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to ask the channel to indicate some sort of condition */
|
|
static int gulp_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session = pvt->session;
|
|
struct ast_sip_session_media *media;
|
|
int response_code = 0;
|
|
int res = 0;
|
|
|
|
switch (condition) {
|
|
case AST_CONTROL_RINGING:
|
|
if (ast_channel_state(ast) == AST_STATE_RING) {
|
|
response_code = 180;
|
|
} else {
|
|
res = -1;
|
|
}
|
|
break;
|
|
case AST_CONTROL_BUSY:
|
|
if (ast_channel_state(ast) != AST_STATE_UP) {
|
|
response_code = 486;
|
|
} else {
|
|
res = -1;
|
|
}
|
|
break;
|
|
case AST_CONTROL_CONGESTION:
|
|
if (ast_channel_state(ast) != AST_STATE_UP) {
|
|
response_code = 503;
|
|
} else {
|
|
res = -1;
|
|
}
|
|
break;
|
|
case AST_CONTROL_INCOMPLETE:
|
|
if (ast_channel_state(ast) != AST_STATE_UP) {
|
|
response_code = 484;
|
|
} else {
|
|
res = -1;
|
|
}
|
|
break;
|
|
case AST_CONTROL_PROCEEDING:
|
|
if (ast_channel_state(ast) != AST_STATE_UP) {
|
|
response_code = 100;
|
|
} else {
|
|
res = -1;
|
|
}
|
|
break;
|
|
case AST_CONTROL_PROGRESS:
|
|
if (ast_channel_state(ast) != AST_STATE_UP) {
|
|
response_code = 183;
|
|
} else {
|
|
res = -1;
|
|
}
|
|
break;
|
|
case AST_CONTROL_VIDUPDATE:
|
|
media = pvt->media[SIP_MEDIA_VIDEO];
|
|
if (media && media->rtp) {
|
|
ast_sip_push_task(session->serializer, transmit_info_with_vidupdate, session);
|
|
} else
|
|
res = -1;
|
|
break;
|
|
case AST_CONTROL_UPDATE_RTP_PEER:
|
|
case AST_CONTROL_PVT_CAUSE_CODE:
|
|
break;
|
|
case AST_CONTROL_HOLD:
|
|
ast_moh_start(ast, data, NULL);
|
|
break;
|
|
case AST_CONTROL_UNHOLD:
|
|
ast_moh_stop(ast);
|
|
break;
|
|
case AST_CONTROL_SRCUPDATE:
|
|
break;
|
|
case AST_CONTROL_SRCCHANGE:
|
|
break;
|
|
case -1:
|
|
res = -1;
|
|
break;
|
|
default:
|
|
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
|
|
res = -1;
|
|
break;
|
|
}
|
|
|
|
if (!res && response_code) {
|
|
struct indicate_data *ind_data = indicate_data_alloc(session, condition, response_code, data, datalen);
|
|
if (ind_data) {
|
|
res = ast_sip_push_task(session->serializer, indicate, ind_data);
|
|
if (res) {
|
|
ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
|
|
response_code, ast_sorcery_object_get_id(session->endpoint));
|
|
ao2_cleanup(ind_data);
|
|
}
|
|
} else {
|
|
res = -1;
|
|
}
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/*! \brief Function called by core to start a DTMF digit */
|
|
static int gulp_digit_begin(struct ast_channel *chan, char digit)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
|
|
struct ast_sip_session *session = pvt->session;
|
|
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
|
|
int res = 0;
|
|
|
|
switch (session->endpoint->dtmf) {
|
|
case AST_SIP_DTMF_RFC_4733:
|
|
if (!media || !media->rtp) {
|
|
return -1;
|
|
}
|
|
|
|
ast_rtp_instance_dtmf_begin(media->rtp, digit);
|
|
case AST_SIP_DTMF_NONE:
|
|
break;
|
|
case AST_SIP_DTMF_INBAND:
|
|
res = -1;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
struct info_dtmf_data {
|
|
struct ast_sip_session *session;
|
|
char digit;
|
|
unsigned int duration;
|
|
};
|
|
|
|
static void info_dtmf_data_destroy(void *obj)
|
|
{
|
|
struct info_dtmf_data *dtmf_data = obj;
|
|
ao2_ref(dtmf_data->session, -1);
|
|
}
|
|
|
|
static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
|
|
{
|
|
struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
|
|
if (!dtmf_data) {
|
|
return NULL;
|
|
}
|
|
ao2_ref(session, +1);
|
|
dtmf_data->session = session;
|
|
dtmf_data->digit = digit;
|
|
dtmf_data->duration = duration;
|
|
return dtmf_data;
|
|
}
|
|
|
|
static int transmit_info_dtmf(void *data)
|
|
{
|
|
RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
|
|
|
|
struct ast_sip_session *session = dtmf_data->session;
|
|
struct pjsip_tx_data *tdata;
|
|
|
|
RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
|
|
|
|
struct ast_sip_body body = {
|
|
.type = "application",
|
|
.subtype = "dtmf-relay",
|
|
};
|
|
|
|
if (!(body_text = ast_str_create(32))) {
|
|
ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
|
|
return -1;
|
|
}
|
|
ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
|
|
|
|
body.body_text = ast_str_buffer(body_text);
|
|
|
|
if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, &tdata)) {
|
|
ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
|
|
return -1;
|
|
}
|
|
if (ast_sip_add_body(tdata, &body)) {
|
|
ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
|
|
pjsip_tx_data_dec_ref(tdata);
|
|
return -1;
|
|
}
|
|
ast_sip_session_send_request(session, tdata);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to stop a DTMF digit */
|
|
static int gulp_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session = pvt->session;
|
|
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
|
|
int res = 0;
|
|
|
|
switch (session->endpoint->dtmf) {
|
|
case AST_SIP_DTMF_INFO:
|
|
{
|
|
struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(session, digit, duration);
|
|
|
|
if (!dtmf_data) {
|
|
return -1;
|
|
}
|
|
|
|
if (ast_sip_push_task(session->serializer, transmit_info_dtmf, dtmf_data)) {
|
|
ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
|
|
ao2_cleanup(dtmf_data);
|
|
return -1;
|
|
}
|
|
break;
|
|
}
|
|
case AST_SIP_DTMF_RFC_4733:
|
|
if (!media || !media->rtp) {
|
|
return -1;
|
|
}
|
|
|
|
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
|
|
case AST_SIP_DTMF_NONE:
|
|
break;
|
|
case AST_SIP_DTMF_INBAND:
|
|
res = -1;
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static int call(void *data)
|
|
{
|
|
pjsip_tx_data *packet;
|
|
struct ast_sip_session *session = data;
|
|
|
|
if (pjsip_inv_invite(session->inv_session, &packet) != PJ_SUCCESS) {
|
|
ast_queue_hangup(session->channel);
|
|
} else {
|
|
ast_sip_session_send_request(session, packet);
|
|
}
|
|
|
|
ao2_ref(session, -1);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to actually start calling a remote party */
|
|
static int gulp_call(struct ast_channel *ast, const char *dest, int timeout)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session = pvt->session;
|
|
|
|
ao2_ref(session, +1);
|
|
if (ast_sip_push_task(session->serializer, call, session)) {
|
|
ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
|
|
ao2_cleanup(session);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
|
|
static int hangup_cause2sip(int cause)
|
|
{
|
|
switch (cause) {
|
|
case AST_CAUSE_UNALLOCATED: /* 1 */
|
|
case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
|
|
case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
|
|
return 404;
|
|
case AST_CAUSE_CONGESTION: /* 34 */
|
|
case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
|
|
return 503;
|
|
case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
|
|
return 408;
|
|
case AST_CAUSE_NO_ANSWER: /* 19 */
|
|
case AST_CAUSE_UNREGISTERED: /* 20 */
|
|
return 480;
|
|
case AST_CAUSE_CALL_REJECTED: /* 21 */
|
|
return 403;
|
|
case AST_CAUSE_NUMBER_CHANGED: /* 22 */
|
|
return 410;
|
|
case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
|
|
return 480;
|
|
case AST_CAUSE_INVALID_NUMBER_FORMAT:
|
|
return 484;
|
|
case AST_CAUSE_USER_BUSY:
|
|
return 486;
|
|
case AST_CAUSE_FAILURE:
|
|
return 500;
|
|
case AST_CAUSE_FACILITY_REJECTED: /* 29 */
|
|
return 501;
|
|
case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
|
|
return 503;
|
|
case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
|
|
return 502;
|
|
case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
|
|
return 488;
|
|
case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
|
|
return 500;
|
|
case AST_CAUSE_NOTDEFINED:
|
|
default:
|
|
ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
|
|
return 0;
|
|
}
|
|
|
|
/* Never reached */
|
|
return 0;
|
|
}
|
|
|
|
struct hangup_data {
|
|
int cause;
|
|
struct ast_channel *chan;
|
|
};
|
|
|
|
static void hangup_data_destroy(void *obj)
|
|
{
|
|
struct hangup_data *h_data = obj;
|
|
|
|
h_data->chan = ast_channel_unref(h_data->chan);
|
|
}
|
|
|
|
static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
|
|
{
|
|
struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
|
|
|
|
if (!h_data) {
|
|
return NULL;
|
|
}
|
|
|
|
h_data->cause = cause;
|
|
h_data->chan = ast_channel_ref(chan);
|
|
|
|
return h_data;
|
|
}
|
|
|
|
static int hangup(void *data)
|
|
{
|
|
pj_status_t status;
|
|
pjsip_tx_data *packet = NULL;
|
|
struct hangup_data *h_data = data;
|
|
struct ast_channel *ast = h_data->chan;
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session = pvt->session;
|
|
int cause = h_data->cause;
|
|
|
|
if (((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS) && packet) {
|
|
if (packet->msg->type == PJSIP_RESPONSE_MSG) {
|
|
ast_sip_session_send_response(session, packet);
|
|
} else {
|
|
ast_sip_session_send_request(session, packet);
|
|
}
|
|
}
|
|
|
|
session->channel = NULL;
|
|
ast_channel_tech_pvt_set(ast, NULL);
|
|
|
|
ao2_cleanup(pvt);
|
|
ao2_cleanup(h_data);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to hang up a Gulp session */
|
|
static int gulp_hangup(struct ast_channel *ast)
|
|
{
|
|
struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session = pvt->session;
|
|
int cause = hangup_cause2sip(ast_channel_hangupcause(session->channel));
|
|
struct hangup_data *h_data = hangup_data_alloc(cause, ast);
|
|
|
|
if (!h_data) {
|
|
goto failure;
|
|
}
|
|
|
|
if (ast_sip_push_task(session->serializer, hangup, h_data)) {
|
|
ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
|
|
goto failure;
|
|
}
|
|
|
|
return 0;
|
|
|
|
failure:
|
|
/* Go ahead and do our cleanup of the session and channel even if we're not going
|
|
* to be able to send our SIP request/response
|
|
*/
|
|
ao2_cleanup(h_data);
|
|
session->channel = NULL;
|
|
ast_channel_tech_pvt_set(ast, NULL);
|
|
|
|
ao2_cleanup(pvt);
|
|
|
|
return -1;
|
|
}
|
|
|
|
struct request_data {
|
|
struct ast_sip_session *session;
|
|
struct ast_format_cap *caps;
|
|
const char *dest;
|
|
int cause;
|
|
};
|
|
|
|
static int request(void *obj)
|
|
{
|
|
struct request_data *req_data = obj;
|
|
struct ast_sip_session *session = NULL;
|
|
char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
|
|
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
|
|
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(endpoint);
|
|
AST_APP_ARG(aor);
|
|
);
|
|
|
|
if (ast_strlen_zero(tmp)) {
|
|
ast_log(LOG_ERROR, "Unable to create Gulp channel with empty destination\n");
|
|
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
|
|
return -1;
|
|
}
|
|
|
|
AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
|
|
|
|
/* If a request user has been specified extract it from the endpoint name portion */
|
|
if ((endpoint_name = strchr(args.endpoint, '@'))) {
|
|
request_user = args.endpoint;
|
|
*endpoint_name++ = '\0';
|
|
} else {
|
|
endpoint_name = args.endpoint;
|
|
}
|
|
|
|
if (ast_strlen_zero(endpoint_name)) {
|
|
ast_log(LOG_ERROR, "Unable to create Gulp channel with empty endpoint name\n");
|
|
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
|
|
} else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
|
|
ast_log(LOG_ERROR, "Unable to create Gulp channel - endpoint '%s' was not found\n", endpoint_name);
|
|
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
return -1;
|
|
}
|
|
|
|
if (!(session = ast_sip_session_create_outgoing(endpoint, args.aor, request_user, req_data->caps))) {
|
|
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
return -1;
|
|
}
|
|
|
|
req_data->session = session;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to create a new outgoing Gulp session */
|
|
static struct ast_channel *gulp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
|
|
{
|
|
struct request_data req_data;
|
|
struct ast_sip_session *session;
|
|
|
|
req_data.caps = cap;
|
|
req_data.dest = data;
|
|
|
|
if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
|
|
*cause = req_data.cause;
|
|
return NULL;
|
|
}
|
|
|
|
session = req_data.session;
|
|
|
|
if (!(session->channel = gulp_new(session, AST_STATE_DOWN, NULL, NULL, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
|
|
/* Session needs to be terminated prematurely */
|
|
return NULL;
|
|
}
|
|
|
|
return session->channel;
|
|
}
|
|
|
|
/*! \brief Function called by core to send text on Gulp session */
|
|
static int gulp_sendtext(struct ast_channel *ast, const char *text)
|
|
{
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
|
|
static int hangup_sip2cause(int cause)
|
|
{
|
|
/* Possible values taken from causes.h */
|
|
|
|
switch(cause) {
|
|
case 401: /* Unauthorized */
|
|
return AST_CAUSE_CALL_REJECTED;
|
|
case 403: /* Not found */
|
|
return AST_CAUSE_CALL_REJECTED;
|
|
case 404: /* Not found */
|
|
return AST_CAUSE_UNALLOCATED;
|
|
case 405: /* Method not allowed */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 407: /* Proxy authentication required */
|
|
return AST_CAUSE_CALL_REJECTED;
|
|
case 408: /* No reaction */
|
|
return AST_CAUSE_NO_USER_RESPONSE;
|
|
case 409: /* Conflict */
|
|
return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
|
|
case 410: /* Gone */
|
|
return AST_CAUSE_NUMBER_CHANGED;
|
|
case 411: /* Length required */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 413: /* Request entity too large */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 414: /* Request URI too large */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 415: /* Unsupported media type */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 420: /* Bad extension */
|
|
return AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
case 480: /* No answer */
|
|
return AST_CAUSE_NO_ANSWER;
|
|
case 481: /* No answer */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 482: /* Loop detected */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 483: /* Too many hops */
|
|
return AST_CAUSE_NO_ANSWER;
|
|
case 484: /* Address incomplete */
|
|
return AST_CAUSE_INVALID_NUMBER_FORMAT;
|
|
case 485: /* Ambiguous */
|
|
return AST_CAUSE_UNALLOCATED;
|
|
case 486: /* Busy everywhere */
|
|
return AST_CAUSE_BUSY;
|
|
case 487: /* Request terminated */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 488: /* No codecs approved */
|
|
return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
|
|
case 491: /* Request pending */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 493: /* Undecipherable */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 500: /* Server internal failure */
|
|
return AST_CAUSE_FAILURE;
|
|
case 501: /* Call rejected */
|
|
return AST_CAUSE_FACILITY_REJECTED;
|
|
case 502:
|
|
return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
|
|
case 503: /* Service unavailable */
|
|
return AST_CAUSE_CONGESTION;
|
|
case 504: /* Gateway timeout */
|
|
return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
|
|
case 505: /* SIP version not supported */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 600: /* Busy everywhere */
|
|
return AST_CAUSE_USER_BUSY;
|
|
case 603: /* Decline */
|
|
return AST_CAUSE_CALL_REJECTED;
|
|
case 604: /* Does not exist anywhere */
|
|
return AST_CAUSE_UNALLOCATED;
|
|
case 606: /* Not acceptable */
|
|
return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
|
|
default:
|
|
if (cause < 500 && cause >= 400) {
|
|
/* 4xx class error that is unknown - someting wrong with our request */
|
|
return AST_CAUSE_INTERWORKING;
|
|
} else if (cause < 600 && cause >= 500) {
|
|
/* 5xx class error - problem in the remote end */
|
|
return AST_CAUSE_CONGESTION;
|
|
} else if (cause < 700 && cause >= 600) {
|
|
/* 6xx - global errors in the 4xx class */
|
|
return AST_CAUSE_INTERWORKING;
|
|
}
|
|
return AST_CAUSE_NORMAL;
|
|
}
|
|
/* Never reached */
|
|
return 0;
|
|
}
|
|
|
|
static void gulp_session_begin(struct ast_sip_session *session)
|
|
{
|
|
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
|
|
|
|
if (session->endpoint->direct_media_glare_mitigation ==
|
|
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
|
|
return;
|
|
}
|
|
|
|
datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
|
|
"direct_media_glare_mitigation");
|
|
|
|
if (!datastore) {
|
|
return;
|
|
}
|
|
|
|
ast_sip_session_add_datastore(session, datastore);
|
|
}
|
|
|
|
/*! \brief Function called when the session ends */
|
|
static void gulp_session_end(struct ast_sip_session *session)
|
|
{
|
|
if (!session->channel) {
|
|
return;
|
|
}
|
|
|
|
if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
|
|
int cause = hangup_sip2cause(session->inv_session->cause);
|
|
|
|
ast_queue_hangup_with_cause(session->channel, cause);
|
|
} else {
|
|
ast_queue_hangup(session->channel);
|
|
}
|
|
}
|
|
|
|
/*! \brief Function called when a request is received on the session */
|
|
static int gulp_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
pjsip_tx_data *packet = NULL;
|
|
int res = AST_PBX_FAILED;
|
|
|
|
if (session->channel) {
|
|
return 0;
|
|
}
|
|
|
|
if (!(session->channel = gulp_new(session, AST_STATE_DOWN, session->exten, NULL, NULL, NULL))) {
|
|
if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
|
|
ast_sip_session_send_response(session, packet);
|
|
}
|
|
|
|
ast_log(LOG_ERROR, "Failed to allocate new GULP channel on incoming SIP INVITE\n");
|
|
return -1;
|
|
}
|
|
|
|
ast_setstate(session->channel, AST_STATE_RING);
|
|
res = ast_pbx_start(session->channel);
|
|
|
|
switch (res) {
|
|
case AST_PBX_FAILED:
|
|
ast_log(LOG_WARNING, "Failed to start PBX ;(\n");
|
|
ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
|
|
ast_hangup(session->channel);
|
|
break;
|
|
case AST_PBX_CALL_LIMIT:
|
|
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
|
|
ast_channel_hangupcause_set(session->channel, AST_CAUSE_SWITCH_CONGESTION);
|
|
ast_hangup(session->channel);
|
|
break;
|
|
case AST_PBX_SUCCESS:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ast_debug(3, "Started PBX on new GULP channel %s\n", ast_channel_name(session->channel));
|
|
|
|
return (res == AST_PBX_SUCCESS) ? 0 : -1;
|
|
}
|
|
|
|
/*! \brief Function called when a response is received on the session */
|
|
static void gulp_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
struct pjsip_status_line status = rdata->msg_info.msg->line.status;
|
|
|
|
if (!session->channel) {
|
|
return;
|
|
}
|
|
|
|
switch (status.code) {
|
|
case 180:
|
|
ast_queue_control(session->channel, AST_CONTROL_RINGING);
|
|
if (ast_channel_state(session->channel) != AST_STATE_UP) {
|
|
ast_setstate(session->channel, AST_STATE_RINGING);
|
|
}
|
|
break;
|
|
case 183:
|
|
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
|
|
break;
|
|
case 200:
|
|
ast_queue_control(session->channel, AST_CONTROL_ANSWER);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
|
|
static int gulp_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
|
|
if (session->endpoint->direct_media) {
|
|
ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \brief Load the module
|
|
*
|
|
* Module loading including tests for configuration or dependencies.
|
|
* This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
|
|
* or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
|
|
* tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
|
|
* configuration file or other non-critical problem return
|
|
* AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
|
|
*/
|
|
static int load_module(void)
|
|
{
|
|
if (!(gulp_tech.capabilities = ast_format_cap_alloc())) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
ast_format_cap_add_all_by_type(gulp_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
|
|
|
|
ast_rtp_glue_register(&gulp_rtp_glue);
|
|
|
|
if (ast_channel_register(&gulp_tech)) {
|
|
ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
|
|
goto end;
|
|
}
|
|
|
|
if (ast_custom_function_register(&gulp_dial_contacts_function)) {
|
|
ast_log(LOG_ERROR, "Unable to register GULP_DIAL_CONTACTS dialplan function\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_sip_session_register_supplement(&gulp_supplement)) {
|
|
ast_log(LOG_ERROR, "Unable to register Gulp supplement\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_sip_session_register_supplement(&gulp_ack_supplement)) {
|
|
ast_log(LOG_ERROR, "Unable to register Gulp ACK supplement\n");
|
|
ast_sip_session_unregister_supplement(&gulp_supplement);
|
|
goto end;
|
|
}
|
|
|
|
return 0;
|
|
|
|
end:
|
|
ast_custom_function_unregister(&gulp_dial_contacts_function);
|
|
ast_channel_unregister(&gulp_tech);
|
|
ast_rtp_glue_unregister(&gulp_rtp_glue);
|
|
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
|
|
/*! \brief Reload module */
|
|
static int reload(void)
|
|
{
|
|
return -1;
|
|
}
|
|
|
|
/*! \brief Unload the Gulp channel from Asterisk */
|
|
static int unload_module(void)
|
|
{
|
|
ast_sip_session_unregister_supplement(&gulp_supplement);
|
|
ast_custom_function_unregister(&gulp_dial_contacts_function);
|
|
ast_channel_unregister(&gulp_tech);
|
|
ast_rtp_glue_unregister(&gulp_rtp_glue);
|
|
|
|
return 0;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Gulp SIP Channel Driver",
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.reload = reload,
|
|
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
|
|
);
|