3431 lines
110 KiB
C
3431 lines
110 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 PSJIP 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_pjsip</depend>
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<depend>res_pjsip_pubsub</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 <pjlib.h>
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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/dsp.h"
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#include "asterisk/stasis_endpoints.h"
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#include "asterisk/stasis_channels.h"
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#include "asterisk/indications.h"
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#include "asterisk/format_cache.h"
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#include "asterisk/translate.h"
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#include "asterisk/threadstorage.h"
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#include "asterisk/features_config.h"
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#include "asterisk/pickup.h"
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#include "asterisk/test.h"
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#include "asterisk/message.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/stream.h"
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#include "pjsip/include/chan_pjsip.h"
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#include "pjsip/include/dialplan_functions.h"
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#include "pjsip/include/cli_functions.h"
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AST_THREADSTORAGE(uniqueid_threadbuf);
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#define UNIQUEID_BUFSIZE 256
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static const char channel_type[] = "PJSIP";
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static unsigned int chan_idx;
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static void chan_pjsip_pvt_dtor(void *obj)
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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 *chan_pjsip_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 *chan_pjsip_request_with_stream_topology(const char *type,
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struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids,
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const struct ast_channel *requestor, const char *data, int *cause);
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static int chan_pjsip_sendtext_data(struct ast_channel *ast, struct ast_msg_data *msg);
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static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text);
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static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit);
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static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
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static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout);
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static int chan_pjsip_hangup(struct ast_channel *ast);
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static int chan_pjsip_answer(struct ast_channel *ast);
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static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast);
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static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f);
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static int chan_pjsip_write_stream(struct ast_channel *ast, int stream_num, struct ast_frame *f);
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static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
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static int chan_pjsip_transfer(struct ast_channel *ast, const char *target);
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static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
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static int chan_pjsip_devicestate(const char *data);
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static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
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static const char *chan_pjsip_get_uniqueid(struct ast_channel *ast);
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/*! \brief PBX interface structure for channel registration */
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struct ast_channel_tech chan_pjsip_tech = {
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.type = channel_type,
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.description = "PJSIP Channel Driver",
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.requester = chan_pjsip_request,
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.requester_with_stream_topology = chan_pjsip_request_with_stream_topology,
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.send_text = chan_pjsip_sendtext,
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.send_text_data = chan_pjsip_sendtext_data,
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.send_digit_begin = chan_pjsip_digit_begin,
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.send_digit_end = chan_pjsip_digit_end,
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.call = chan_pjsip_call,
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.hangup = chan_pjsip_hangup,
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.answer = chan_pjsip_answer,
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.read_stream = chan_pjsip_read_stream,
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.write = chan_pjsip_write,
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.write_stream = chan_pjsip_write_stream,
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.exception = chan_pjsip_read_stream,
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.indicate = chan_pjsip_indicate,
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.transfer = chan_pjsip_transfer,
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.fixup = chan_pjsip_fixup,
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.devicestate = chan_pjsip_devicestate,
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.queryoption = chan_pjsip_queryoption,
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.func_channel_read = pjsip_acf_channel_read,
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.get_pvt_uniqueid = chan_pjsip_get_uniqueid,
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.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER | AST_CHAN_TP_SEND_TEXT_DATA
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};
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/*! \brief SIP session interaction functions */
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static void chan_pjsip_session_begin(struct ast_sip_session *session);
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static void chan_pjsip_session_end(struct ast_sip_session *session);
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static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
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static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
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static void chan_pjsip_incoming_response_update_cause(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 chan_pjsip_supplement = {
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.method = "INVITE",
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.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
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.session_begin = chan_pjsip_session_begin,
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.session_end = chan_pjsip_session_end,
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.incoming_request = chan_pjsip_incoming_request,
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.incoming_response = chan_pjsip_incoming_response,
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/* It is important that this supplement runs after media has been negotiated */
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.response_priority = AST_SIP_SESSION_AFTER_MEDIA,
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};
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/*! \brief SIP session supplement structure just for responses */
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static struct ast_sip_session_supplement chan_pjsip_supplement_response = {
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.method = "INVITE",
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.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
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.incoming_response = chan_pjsip_incoming_response_update_cause,
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.response_priority = AST_SIP_SESSION_BEFORE_MEDIA | AST_SIP_SESSION_AFTER_MEDIA,
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};
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static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
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static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
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.method = "ACK",
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.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
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.incoming_request = chan_pjsip_incoming_ack,
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};
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static int chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
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static struct ast_sip_session_supplement chan_pjsip_prack_supplement = {
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.method = "PRACK",
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.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
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.incoming_request = chan_pjsip_incoming_prack,
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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 chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
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{
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struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
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struct ast_sip_endpoint *endpoint;
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struct ast_datastore *datastore;
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struct ast_sip_session_media *media;
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if (!channel || !channel->session) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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/* XXX Getting the first RTP instance for direct media related stuff seems just
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* absolutely wrong. But the native RTP bridge knows no other method than single-stream
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* for direct media. So this is the best we can do.
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*/
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media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
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if (!media || !media->rtp) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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datastore = ast_sip_session_get_datastore(channel->session, "t38");
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if (datastore) {
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ao2_ref(datastore, -1);
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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endpoint = channel->session->endpoint;
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*instance = media->rtp;
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ao2_ref(*instance, +1);
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ast_assert(endpoint != NULL);
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if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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if (endpoint->media.direct_media.enabled) {
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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 chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
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{
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struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
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struct ast_sip_endpoint *endpoint;
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struct ast_sip_session_media *media;
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if (!channel || !channel->session) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO];
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if (!media || !media->rtp) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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endpoint = channel->session->endpoint;
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*instance = media->rtp;
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ao2_ref(*instance, +1);
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ast_assert(endpoint != NULL);
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if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
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return AST_RTP_GLUE_RESULT_FORBID;
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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 peer capabilities */
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static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
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{
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SCOPE_ENTER(1, "%s Native formats %s\n", ast_channel_name(chan),
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ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(ast_channel_nativeformats(chan), &STR_TMP)));
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ast_format_cap_append_from_cap(result, ast_channel_nativeformats(chan), AST_MEDIA_TYPE_UNKNOWN);
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SCOPE_EXIT_RTN();
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}
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/*! \brief Destructor function for \ref transport_info_data */
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static void transport_info_destroy(void *obj)
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{
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struct transport_info_data *data = obj;
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ast_free(data);
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}
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/*! \brief Datastore used to store local/remote addresses for the
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* INVITE request that created the PJSIP channel */
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static struct ast_datastore_info transport_info = {
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.type = "chan_pjsip_transport_info",
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.destroy = transport_info_destroy,
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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->media.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->media.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->media.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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/*! \brief Helper function to find the position for RTCP */
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static int rtp_find_rtcp_fd_position(struct ast_sip_session *session, struct ast_rtp_instance *rtp)
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{
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int index;
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for (index = 0; index < AST_VECTOR_SIZE(&session->active_media_state->read_callbacks); ++index) {
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struct ast_sip_session_media_read_callback_state *callback_state =
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AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, index);
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if (callback_state->fd != ast_rtp_instance_fd(rtp, 1)) {
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continue;
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}
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return index;
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}
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return -1;
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}
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/*!
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* \pre chan is locked
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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, struct ast_sip_session *session)
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{
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int changed = 0, position = -1;
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if (media->rtp) {
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position = rtp_find_rtcp_fd_position(session, media->rtp);
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}
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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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if (position != -1) {
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ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, -1);
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}
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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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if (position != -1) {
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ast_channel_set_fd(chan, position + AST_EXTENDED_FDS, ast_rtp_instance_fd(media->rtp, 1));
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}
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}
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}
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return changed;
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}
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struct rtp_direct_media_data {
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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_format_cap *cap;
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struct ast_sip_session *session;
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};
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static void rtp_direct_media_data_destroy(void *data)
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{
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struct rtp_direct_media_data *cdata = data;
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ao2_cleanup(cdata->session);
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ao2_cleanup(cdata->cap);
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ao2_cleanup(cdata->vrtp);
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ao2_cleanup(cdata->rtp);
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ao2_cleanup(cdata->chan);
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}
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static struct rtp_direct_media_data *rtp_direct_media_data_create(
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struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp,
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const struct ast_format_cap *cap, struct ast_sip_session *session)
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{
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struct rtp_direct_media_data *cdata = ao2_alloc(sizeof(*cdata), rtp_direct_media_data_destroy);
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if (!cdata) {
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return NULL;
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}
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cdata->chan = ao2_bump(chan);
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cdata->rtp = ao2_bump(rtp);
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cdata->vrtp = ao2_bump(vrtp);
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cdata->cap = ao2_bump((struct ast_format_cap *)cap);
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cdata->session = ao2_bump(session);
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return cdata;
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}
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static int send_direct_media_request(void *data)
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{
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struct rtp_direct_media_data *cdata = data;
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struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(cdata->chan);
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struct ast_sip_session *session;
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int changed = 0;
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int res = 0;
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/* XXX In an ideal world each media stream would be direct, but for now preserve behavior
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* and connect only the default media sessions for audio and video.
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*/
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/* The channel needs to be locked when checking for RTP changes.
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* Otherwise, we could end up destroying an underlying RTCP structure
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* at the same time that the channel thread is attempting to read RTCP
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*/
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ast_channel_lock(cdata->chan);
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session = channel->session;
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if (session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
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changed |= check_for_rtp_changes(
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cdata->chan, cdata->rtp, session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO], session);
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}
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if (session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO]) {
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changed |= check_for_rtp_changes(
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cdata->chan, cdata->vrtp, session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO], session);
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}
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ast_channel_unlock(cdata->chan);
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if (direct_media_mitigate_glare(cdata->session)) {
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ast_debug(4, "Disregarding setting RTP on %s: mitigating re-INVITE glare\n", ast_channel_name(cdata->chan));
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ao2_ref(cdata, -1);
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return 0;
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}
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if (cdata->cap && ast_format_cap_count(cdata->cap) &&
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!ast_format_cap_identical(cdata->session->direct_media_cap, cdata->cap)) {
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|
ast_format_cap_remove_by_type(cdata->session->direct_media_cap, AST_MEDIA_TYPE_UNKNOWN);
|
|
ast_format_cap_append_from_cap(cdata->session->direct_media_cap, cdata->cap, AST_MEDIA_TYPE_UNKNOWN);
|
|
changed = 1;
|
|
}
|
|
|
|
if (changed) {
|
|
ast_debug(4, "RTP changed on %s; initiating direct media update\n", ast_channel_name(cdata->chan));
|
|
res = ast_sip_session_refresh(cdata->session, NULL, NULL, NULL,
|
|
cdata->session->endpoint->media.direct_media.method, 1, NULL);
|
|
}
|
|
|
|
ao2_ref(cdata, -1);
|
|
return res;
|
|
}
|
|
|
|
/*! \brief Function called by RTP engine to change where the remote party should send media */
|
|
static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
|
|
struct ast_rtp_instance *rtp,
|
|
struct ast_rtp_instance *vrtp,
|
|
struct ast_rtp_instance *tpeer,
|
|
const struct ast_format_cap *cap,
|
|
int nat_active)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
|
|
struct ast_sip_session *session = channel->session;
|
|
struct rtp_direct_media_data *cdata;
|
|
SCOPE_ENTER(1, "%s %s\n", ast_channel_name(chan),
|
|
ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(cap, &STR_TMP)));
|
|
|
|
/* Don't try to do any direct media shenanigans on early bridges */
|
|
if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) {
|
|
ast_debug(4, "Disregarding setting RTP on %s: channel is not bridged\n", ast_channel_name(chan));
|
|
SCOPE_EXIT_RTN_VALUE(0, "Channel not bridged\n");
|
|
}
|
|
|
|
if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
|
|
ast_debug(4, "Disregarding setting RTP on %s: NAT is active\n", ast_channel_name(chan));
|
|
SCOPE_EXIT_RTN_VALUE(0, "NAT is active\n");
|
|
}
|
|
|
|
cdata = rtp_direct_media_data_create(chan, rtp, vrtp, cap, session);
|
|
if (!cdata) {
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
if (ast_sip_push_task(session->serializer, send_direct_media_request, cdata)) {
|
|
ast_log(LOG_ERROR, "Unable to send direct media request for channel %s\n", ast_channel_name(chan));
|
|
ao2_ref(cdata, -1);
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
/*! \brief Local glue for interacting with the RTP engine core */
|
|
static struct ast_rtp_glue chan_pjsip_rtp_glue = {
|
|
.type = "PJSIP",
|
|
.get_rtp_info = chan_pjsip_get_rtp_peer,
|
|
.get_vrtp_info = chan_pjsip_get_vrtp_peer,
|
|
.get_codec = chan_pjsip_get_codec,
|
|
.update_peer = chan_pjsip_set_rtp_peer,
|
|
};
|
|
|
|
static void set_channel_on_rtp_instance(const struct ast_sip_session *session,
|
|
const char *channel_id)
|
|
{
|
|
int i;
|
|
|
|
for (i = 0; i < AST_VECTOR_SIZE(&session->active_media_state->sessions); ++i) {
|
|
struct ast_sip_session_media *session_media;
|
|
|
|
session_media = AST_VECTOR_GET(&session->active_media_state->sessions, i);
|
|
if (!session_media || !session_media->rtp) {
|
|
continue;
|
|
}
|
|
|
|
ast_rtp_instance_set_channel_id(session_media->rtp, channel_id);
|
|
}
|
|
}
|
|
|
|
/*!
|
|
* \brief Determine if a topology is compatible with format capabilities
|
|
*
|
|
* This will return true if ANY formats in the topology are compatible with the format
|
|
* capabilities.
|
|
*
|
|
* XXX When supporting true multistream, we will need to be sure to mark which streams from
|
|
* top1 are compatible with which streams from top2. Then the ones that are not compatible
|
|
* will need to be marked as "removed" so that they are negotiated as expected.
|
|
*
|
|
* \param top Topology
|
|
* \param cap Format capabilities
|
|
* \retval 1 The topology has at least one compatible format
|
|
* \retval 0 The topology has no compatible formats or an error occurred.
|
|
*/
|
|
static int compatible_formats_exist(struct ast_stream_topology *top, struct ast_format_cap *cap)
|
|
{
|
|
struct ast_format_cap *cap_from_top;
|
|
int res;
|
|
SCOPE_ENTER(1, "Topology: %s Formats: %s\n",
|
|
ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_stream_topology_to_str(top, &STR_TMP)),
|
|
ast_str_tmp(AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_get_names(cap, &STR_TMP)));
|
|
|
|
cap_from_top = ast_stream_topology_get_formats(top);
|
|
|
|
if (!cap_from_top) {
|
|
SCOPE_EXIT_RTN_VALUE(0, "Topology had no formats\n");
|
|
}
|
|
|
|
res = ast_format_cap_iscompatible(cap_from_top, cap);
|
|
ao2_ref(cap_from_top, -1);
|
|
|
|
SCOPE_EXIT_RTN_VALUE(res, "Compatible? %s\n", res ? "yes" : "no");
|
|
}
|
|
|
|
/*! \brief Function called to create a new PJSIP Asterisk channel */
|
|
static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name)
|
|
{
|
|
struct ast_channel *chan;
|
|
struct ast_format_cap *caps;
|
|
RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
|
|
struct ast_sip_channel_pvt *channel;
|
|
struct ast_variable *var;
|
|
struct ast_stream_topology *topology;
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (!(pvt = ao2_alloc_options(sizeof(*pvt), chan_pjsip_pvt_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK))) {
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt\n");
|
|
}
|
|
|
|
chan = ast_channel_alloc_with_endpoint(1, state,
|
|
S_COR(session->id.number.valid, session->id.number.str, ""),
|
|
S_COR(session->id.name.valid, session->id.name.str, ""),
|
|
session->endpoint->accountcode,
|
|
exten, session->endpoint->context,
|
|
assignedids, requestor, 0,
|
|
session->endpoint->persistent, "PJSIP/%s-%08x",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
(unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
|
|
if (!chan) {
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
|
|
}
|
|
|
|
ast_channel_tech_set(chan, &chan_pjsip_tech);
|
|
|
|
if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
|
|
ast_channel_unlock(chan);
|
|
ast_hangup(chan);
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create pvt channel\n");
|
|
}
|
|
|
|
ast_channel_tech_pvt_set(chan, channel);
|
|
|
|
if (!ast_stream_topology_get_count(session->pending_media_state->topology) ||
|
|
!compatible_formats_exist(session->pending_media_state->topology, session->endpoint->media.codecs)) {
|
|
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
|
|
if (!caps) {
|
|
ast_channel_unlock(chan);
|
|
ast_hangup(chan);
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create caps\n");
|
|
}
|
|
ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
|
|
topology = ast_stream_topology_clone(session->endpoint->media.topology);
|
|
} else {
|
|
caps = ast_stream_topology_get_formats(session->pending_media_state->topology);
|
|
topology = ast_stream_topology_clone(session->pending_media_state->topology);
|
|
}
|
|
|
|
if (!topology || !caps) {
|
|
ao2_cleanup(caps);
|
|
ast_stream_topology_free(topology);
|
|
ast_channel_unlock(chan);
|
|
ast_hangup(chan);
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't get caps or clone topology\n");
|
|
}
|
|
|
|
ast_channel_stage_snapshot(chan);
|
|
|
|
ast_channel_nativeformats_set(chan, caps);
|
|
ast_channel_set_stream_topology(chan, topology);
|
|
|
|
if (!ast_format_cap_empty(caps)) {
|
|
struct ast_format *fmt;
|
|
|
|
fmt = ast_format_cap_get_best_by_type(caps, AST_MEDIA_TYPE_AUDIO);
|
|
if (!fmt) {
|
|
/* Since our capabilities aren't empty, this will succeed */
|
|
fmt = ast_format_cap_get_format(caps, 0);
|
|
}
|
|
ast_channel_set_writeformat(chan, fmt);
|
|
ast_channel_set_rawwriteformat(chan, fmt);
|
|
ast_channel_set_readformat(chan, fmt);
|
|
ast_channel_set_rawreadformat(chan, fmt);
|
|
ao2_ref(fmt, -1);
|
|
}
|
|
|
|
ao2_ref(caps, -1);
|
|
|
|
if (state == AST_STATE_RING) {
|
|
ast_channel_rings_set(chan, 1);
|
|
}
|
|
|
|
ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
|
|
|
|
ast_party_id_copy(&ast_channel_caller(chan)->id, &session->id);
|
|
ast_party_id_copy(&ast_channel_caller(chan)->ani, &session->id);
|
|
ast_channel_caller(chan)->ani2 = session->ani2;
|
|
|
|
if (!ast_strlen_zero(exten)) {
|
|
/* Set provided DNID on the new channel. */
|
|
ast_channel_dialed(chan)->number.str = ast_strdup(exten);
|
|
}
|
|
|
|
ast_channel_priority_set(chan, 1);
|
|
|
|
ast_channel_callgroup_set(chan, session->endpoint->pickup.callgroup);
|
|
ast_channel_pickupgroup_set(chan, session->endpoint->pickup.pickupgroup);
|
|
|
|
ast_channel_named_callgroups_set(chan, session->endpoint->pickup.named_callgroups);
|
|
ast_channel_named_pickupgroups_set(chan, session->endpoint->pickup.named_pickupgroups);
|
|
|
|
if (!ast_strlen_zero(session->endpoint->language)) {
|
|
ast_channel_language_set(chan, session->endpoint->language);
|
|
}
|
|
|
|
if (!ast_strlen_zero(session->endpoint->zone)) {
|
|
struct ast_tone_zone *zone = ast_get_indication_zone(session->endpoint->zone);
|
|
if (!zone) {
|
|
ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", session->endpoint->zone);
|
|
}
|
|
ast_channel_zone_set(chan, zone);
|
|
}
|
|
|
|
for (var = session->endpoint->channel_vars; var; var = var->next) {
|
|
char buf[512];
|
|
pbx_builtin_setvar_helper(chan, var->name, ast_get_encoded_str(
|
|
var->value, buf, sizeof(buf)));
|
|
}
|
|
|
|
ast_channel_stage_snapshot_done(chan);
|
|
ast_channel_unlock(chan);
|
|
|
|
set_channel_on_rtp_instance(session, ast_channel_uniqueid(chan));
|
|
|
|
SCOPE_EXIT_RTN_VALUE(chan);
|
|
}
|
|
|
|
struct answer_data {
|
|
struct ast_sip_session *session;
|
|
unsigned long indent;
|
|
};
|
|
|
|
static int answer(void *data)
|
|
{
|
|
struct answer_data *ans_data = data;
|
|
pj_status_t status = PJ_SUCCESS;
|
|
pjsip_tx_data *packet = NULL;
|
|
struct ast_sip_session *session = ans_data->session;
|
|
SCOPE_ENTER_TASK(1, ans_data->indent, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
|
|
session->inv_session->cause,
|
|
pjsip_get_status_text(session->inv_session->cause)->ptr);
|
|
SCOPE_EXIT_RTN_VALUE(0, "Disconnected\n");
|
|
}
|
|
|
|
pjsip_dlg_inc_lock(session->inv_session->dlg);
|
|
if (session->inv_session->invite_tsx) {
|
|
status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet);
|
|
} else {
|
|
ast_log(LOG_ERROR,"Cannot answer '%s' because there is no associated SIP transaction\n",
|
|
ast_channel_name(session->channel));
|
|
}
|
|
pjsip_dlg_dec_lock(session->inv_session->dlg);
|
|
|
|
if (status == PJ_SUCCESS && packet) {
|
|
ast_sip_session_send_response(session, packet);
|
|
}
|
|
|
|
if (status != PJ_SUCCESS) {
|
|
char err[PJ_ERR_MSG_SIZE];
|
|
|
|
pj_strerror(status, err, sizeof(err));
|
|
ast_log(LOG_WARNING,"Cannot answer '%s': %s\n",
|
|
ast_channel_name(session->channel), err);
|
|
/*
|
|
* Return this value so we can distinguish between this
|
|
* failure and the threadpool synchronous push failing.
|
|
*/
|
|
SCOPE_EXIT_RTN_VALUE(-2, "pjproject failure\n");
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
/*! \brief Function called by core when we should answer a PJSIP session */
|
|
static int chan_pjsip_answer(struct ast_channel *ast)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session;
|
|
struct answer_data ans_data = { 0, };
|
|
int res;
|
|
SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
|
|
|
|
if (ast_channel_state(ast) == AST_STATE_UP) {
|
|
SCOPE_EXIT_RTN_VALUE(0, "Already up\n");
|
|
return 0;
|
|
}
|
|
|
|
ast_setstate(ast, AST_STATE_UP);
|
|
session = ao2_bump(channel->session);
|
|
|
|
/* the answer task needs to be pushed synchronously otherwise a race condition
|
|
can occur between this thread and bridging (specifically when native bridging
|
|
attempts to do direct media) */
|
|
ast_channel_unlock(ast);
|
|
ans_data.session = session;
|
|
ans_data.indent = ast_trace_get_indent();
|
|
res = ast_sip_push_task_wait_serializer(session->serializer, answer, &ans_data);
|
|
if (res) {
|
|
if (res == -1) {
|
|
ast_log(LOG_ERROR,"Cannot answer '%s': Unable to push answer task to the threadpool.\n",
|
|
ast_channel_name(session->channel));
|
|
}
|
|
ao2_ref(session, -1);
|
|
ast_channel_lock(ast);
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
|
|
}
|
|
ao2_ref(session, -1);
|
|
ast_channel_lock(ast);
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
/*! \brief Internal helper function called when CNG tone is detected */
|
|
static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_channel *ast, struct ast_sip_session *session,
|
|
struct ast_frame *f)
|
|
{
|
|
const char *target_context;
|
|
int exists;
|
|
int dsp_features;
|
|
|
|
dsp_features = ast_dsp_get_features(session->dsp);
|
|
dsp_features &= ~DSP_FEATURE_FAX_DETECT;
|
|
if (dsp_features) {
|
|
ast_dsp_set_features(session->dsp, dsp_features);
|
|
} else {
|
|
ast_dsp_free(session->dsp);
|
|
session->dsp = NULL;
|
|
}
|
|
|
|
/* If already executing in the fax extension don't do anything */
|
|
if (!strcmp(ast_channel_exten(ast), "fax")) {
|
|
return f;
|
|
}
|
|
|
|
target_context = ast_channel_context(ast);
|
|
|
|
/*
|
|
* We need to unlock the channel here because ast_exists_extension has the
|
|
* potential to start and stop an autoservice on the channel. Such action
|
|
* is prone to deadlock if the channel is locked.
|
|
*
|
|
* ast_async_goto() has its own restriction on not holding the channel lock.
|
|
*/
|
|
ast_channel_unlock(ast);
|
|
ast_frfree(f);
|
|
f = &ast_null_frame;
|
|
exists = ast_exists_extension(ast, target_context, "fax", 1,
|
|
S_COR(ast_channel_caller(ast)->id.number.valid,
|
|
ast_channel_caller(ast)->id.number.str, NULL));
|
|
if (exists) {
|
|
ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
|
|
ast_channel_name(ast));
|
|
pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
|
|
if (ast_async_goto(ast, target_context, "fax", 1)) {
|
|
ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
|
|
ast_channel_name(ast), target_context);
|
|
}
|
|
} else {
|
|
ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
|
|
ast_channel_name(ast), target_context);
|
|
}
|
|
|
|
/* It's possible for a masquerade to have occurred when doing the ast_async_goto resulting in
|
|
* the channel on the session having changed. Since we need to return with the original channel
|
|
* locked we lock the channel that was passed in and not session->channel.
|
|
*/
|
|
ast_channel_lock(ast);
|
|
|
|
return f;
|
|
}
|
|
|
|
/*! \brief Determine if the given frame is in a format we've negotiated */
|
|
static int is_compatible_format(struct ast_sip_session *session, struct ast_frame *f)
|
|
{
|
|
struct ast_stream_topology *topology = session->active_media_state->topology;
|
|
struct ast_stream *stream = ast_stream_topology_get_stream(topology, f->stream_num);
|
|
const struct ast_format_cap *cap = ast_stream_get_formats(stream);
|
|
|
|
return ast_format_cap_iscompatible_format(cap, f->subclass.format) != AST_FORMAT_CMP_NOT_EQUAL;
|
|
}
|
|
|
|
/*!
|
|
* \brief Function called by core to read any waiting frames
|
|
*
|
|
* \note The channel is already locked.
|
|
*/
|
|
static struct ast_frame *chan_pjsip_read_stream(struct ast_channel *ast)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session = channel->session;
|
|
struct ast_sip_session_media_read_callback_state *callback_state;
|
|
struct ast_frame *f;
|
|
int fdno = ast_channel_fdno(ast) - AST_EXTENDED_FDS;
|
|
struct ast_frame *cur;
|
|
|
|
if (fdno >= AST_VECTOR_SIZE(&session->active_media_state->read_callbacks)) {
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
callback_state = AST_VECTOR_GET_ADDR(&session->active_media_state->read_callbacks, fdno);
|
|
f = callback_state->read_callback(session, callback_state->session);
|
|
|
|
if (!f) {
|
|
return f;
|
|
}
|
|
|
|
for (cur = f; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
|
|
if (cur->frametype == AST_FRAME_VOICE) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (!cur || callback_state->session != session->active_media_state->default_session[callback_state->session->type]) {
|
|
return f;
|
|
}
|
|
|
|
session = channel->session;
|
|
|
|
/*
|
|
* Asymmetric RTP only has one native format set at a time.
|
|
* Therefore we need to update the native format to the current
|
|
* raw read format BEFORE the native format check
|
|
*/
|
|
if (!session->endpoint->asymmetric_rtp_codec &&
|
|
ast_format_cmp(ast_channel_rawwriteformat(ast), cur->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL &&
|
|
is_compatible_format(session, cur)) {
|
|
struct ast_format_cap *caps;
|
|
|
|
/* For maximum compatibility we ensure that the formats match that of the received media */
|
|
ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when we're sending '%s', switching to match\n",
|
|
ast_format_get_name(cur->subclass.format), ast_channel_name(ast),
|
|
ast_format_get_name(ast_channel_rawwriteformat(ast)));
|
|
|
|
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
|
|
if (caps) {
|
|
ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(ast), AST_MEDIA_TYPE_UNKNOWN);
|
|
ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_AUDIO);
|
|
ast_format_cap_append(caps, cur->subclass.format, 0);
|
|
ast_channel_nativeformats_set(ast, caps);
|
|
ao2_ref(caps, -1);
|
|
}
|
|
|
|
ast_set_write_format_path(ast, ast_channel_writeformat(ast), cur->subclass.format);
|
|
ast_set_read_format_path(ast, ast_channel_readformat(ast), cur->subclass.format);
|
|
|
|
if (ast_channel_is_bridged(ast)) {
|
|
ast_channel_set_unbridged_nolock(ast, 1);
|
|
}
|
|
}
|
|
|
|
if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), cur->subclass.format)
|
|
== AST_FORMAT_CMP_NOT_EQUAL) {
|
|
ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when it has not been negotiated\n",
|
|
ast_format_get_name(cur->subclass.format), ast_channel_name(ast));
|
|
ast_frfree(f);
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
if (session->dsp) {
|
|
int dsp_features;
|
|
|
|
dsp_features = ast_dsp_get_features(session->dsp);
|
|
if ((dsp_features & DSP_FEATURE_FAX_DETECT)
|
|
&& session->endpoint->faxdetect_timeout
|
|
&& session->endpoint->faxdetect_timeout <= ast_channel_get_up_time(ast)) {
|
|
dsp_features &= ~DSP_FEATURE_FAX_DETECT;
|
|
if (dsp_features) {
|
|
ast_dsp_set_features(session->dsp, dsp_features);
|
|
} else {
|
|
ast_dsp_free(session->dsp);
|
|
session->dsp = NULL;
|
|
}
|
|
ast_debug(3, "Channel driver fax CNG detection timeout on %s\n",
|
|
ast_channel_name(ast));
|
|
}
|
|
}
|
|
if (session->dsp) {
|
|
f = ast_dsp_process(ast, session->dsp, f);
|
|
if (f && (f->frametype == AST_FRAME_DTMF)) {
|
|
if (f->subclass.integer == 'f') {
|
|
ast_debug(3, "Channel driver fax CNG detected on %s\n",
|
|
ast_channel_name(ast));
|
|
f = chan_pjsip_cng_tone_detected(ast, session, f);
|
|
/* When chan_pjsip_cng_tone_detected returns it is possible for the
|
|
* channel pointed to by ast and by session->channel to differ due to a
|
|
* masquerade. It's best not to touch things after this.
|
|
*/
|
|
} else {
|
|
ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
|
|
ast_channel_name(ast));
|
|
}
|
|
}
|
|
}
|
|
|
|
return f;
|
|
}
|
|
|
|
static int chan_pjsip_write_stream(struct ast_channel *ast, int stream_num, struct ast_frame *frame)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session *session = channel->session;
|
|
struct ast_sip_session_media *media = NULL;
|
|
int res = 0;
|
|
|
|
/* The core provides a guarantee that the stream will exist when we are called if stream_num is provided */
|
|
if (stream_num >= 0) {
|
|
/* What is not guaranteed is that a media session will exist */
|
|
if (stream_num < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions)) {
|
|
media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, stream_num);
|
|
}
|
|
}
|
|
|
|
switch (frame->frametype) {
|
|
case AST_FRAME_VOICE:
|
|
if (!media) {
|
|
return 0;
|
|
} else if (media->type != AST_MEDIA_TYPE_AUDIO) {
|
|
ast_debug(3, "Channel %s stream %d is of type '%s', not audio!\n",
|
|
ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
|
|
return 0;
|
|
} else if (media == channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO] &&
|
|
ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
|
|
struct ast_str *cap_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
|
|
struct ast_str *write_transpath = ast_str_alloca(256);
|
|
struct ast_str *read_transpath = ast_str_alloca(256);
|
|
|
|
ast_log(LOG_WARNING,
|
|
"Channel %s asked to send %s frame when native formats are %s (rd:%s->%s;%s wr:%s->%s;%s)\n",
|
|
ast_channel_name(ast),
|
|
ast_format_get_name(frame->subclass.format),
|
|
ast_format_cap_get_names(ast_channel_nativeformats(ast), &cap_buf),
|
|
ast_format_get_name(ast_channel_rawreadformat(ast)),
|
|
ast_format_get_name(ast_channel_readformat(ast)),
|
|
ast_translate_path_to_str(ast_channel_readtrans(ast), &read_transpath),
|
|
ast_format_get_name(ast_channel_writeformat(ast)),
|
|
ast_format_get_name(ast_channel_rawwriteformat(ast)),
|
|
ast_translate_path_to_str(ast_channel_writetrans(ast), &write_transpath));
|
|
return 0;
|
|
} else if (media->write_callback) {
|
|
res = media->write_callback(session, media, frame);
|
|
|
|
}
|
|
break;
|
|
case AST_FRAME_VIDEO:
|
|
if (!media) {
|
|
return 0;
|
|
} else if (media->type != AST_MEDIA_TYPE_VIDEO) {
|
|
ast_debug(3, "Channel %s stream %d is of type '%s', not video!\n",
|
|
ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
|
|
return 0;
|
|
} else if (media->write_callback) {
|
|
res = media->write_callback(session, media, frame);
|
|
}
|
|
break;
|
|
case AST_FRAME_MODEM:
|
|
if (!media) {
|
|
return 0;
|
|
} else if (media->type != AST_MEDIA_TYPE_IMAGE) {
|
|
ast_debug(3, "Channel %s stream %d is of type '%s', not image!\n",
|
|
ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
|
|
return 0;
|
|
} else if (media->write_callback) {
|
|
res = media->write_callback(session, media, frame);
|
|
}
|
|
break;
|
|
case AST_FRAME_CNG:
|
|
break;
|
|
case AST_FRAME_RTCP:
|
|
/* We only support writing out feedback */
|
|
if (frame->subclass.integer != AST_RTP_RTCP_PSFB || !media) {
|
|
return 0;
|
|
} else if (media->type != AST_MEDIA_TYPE_VIDEO) {
|
|
ast_debug(3, "Channel %s stream %d is of type '%s', not video! Unable to write RTCP feedback.\n",
|
|
ast_channel_name(ast), stream_num, ast_codec_media_type2str(media->type));
|
|
return 0;
|
|
} else if (media->write_callback) {
|
|
res = media->write_callback(session, media, frame);
|
|
}
|
|
break;
|
|
default:
|
|
ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP\n", frame->frametype);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
|
|
{
|
|
return chan_pjsip_write_stream(ast, -1, frame);
|
|
}
|
|
|
|
/*! \brief Function called by core to change the underlying owner channel */
|
|
static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
|
|
|
|
if (channel->session->channel != oldchan) {
|
|
return -1;
|
|
}
|
|
|
|
/*
|
|
* The masquerade has suspended the channel's session
|
|
* serializer so we can safely change it outside of
|
|
* the serializer thread.
|
|
*/
|
|
channel->session->channel = newchan;
|
|
|
|
set_channel_on_rtp_instance(channel->session, ast_channel_uniqueid(newchan));
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! AO2 hash function for on hold UIDs */
|
|
static int uid_hold_hash_fn(const void *obj, const int flags)
|
|
{
|
|
const char *key = obj;
|
|
|
|
switch (flags & OBJ_SEARCH_MASK) {
|
|
case OBJ_SEARCH_KEY:
|
|
break;
|
|
case OBJ_SEARCH_OBJECT:
|
|
break;
|
|
default:
|
|
/* Hash can only work on something with a full key. */
|
|
ast_assert(0);
|
|
return 0;
|
|
}
|
|
return ast_str_hash(key);
|
|
}
|
|
|
|
/*! AO2 sort function for on hold UIDs */
|
|
static int uid_hold_sort_fn(const void *obj_left, const void *obj_right, const int flags)
|
|
{
|
|
const char *left = obj_left;
|
|
const char *right = obj_right;
|
|
int cmp;
|
|
|
|
switch (flags & OBJ_SEARCH_MASK) {
|
|
case OBJ_SEARCH_OBJECT:
|
|
case OBJ_SEARCH_KEY:
|
|
cmp = strcmp(left, right);
|
|
break;
|
|
case OBJ_SEARCH_PARTIAL_KEY:
|
|
cmp = strncmp(left, right, strlen(right));
|
|
break;
|
|
default:
|
|
/* Sort can only work on something with a full or partial key. */
|
|
ast_assert(0);
|
|
cmp = 0;
|
|
break;
|
|
}
|
|
return cmp;
|
|
}
|
|
|
|
static struct ao2_container *pjsip_uids_onhold;
|
|
|
|
/*!
|
|
* \brief Add a channel ID to the list of PJSIP channels on hold
|
|
*
|
|
* \param chan_uid - Unique ID of the channel being put into the hold list
|
|
*
|
|
* \retval 0 Channel has been added to or was already in the hold list
|
|
* \retval -1 Failed to add channel to the hold list
|
|
*/
|
|
static int chan_pjsip_add_hold(const char *chan_uid)
|
|
{
|
|
RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
|
|
|
|
hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
|
|
if (hold_uid) {
|
|
/* Device is already on hold. Nothing to do. */
|
|
return 0;
|
|
}
|
|
|
|
/* Device wasn't in hold list already. Create a new one. */
|
|
hold_uid = ao2_alloc_options(strlen(chan_uid) + 1, NULL,
|
|
AO2_ALLOC_OPT_LOCK_NOLOCK);
|
|
if (!hold_uid) {
|
|
return -1;
|
|
}
|
|
|
|
ast_copy_string(hold_uid, chan_uid, strlen(chan_uid) + 1);
|
|
|
|
if (ao2_link(pjsip_uids_onhold, hold_uid) == 0) {
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \brief Remove a channel ID from the list of PJSIP channels on hold
|
|
*
|
|
* \param chan_uid - Unique ID of the channel being taken out of the hold list
|
|
*/
|
|
static void chan_pjsip_remove_hold(const char *chan_uid)
|
|
{
|
|
ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY | OBJ_UNLINK | OBJ_NODATA);
|
|
}
|
|
|
|
/*!
|
|
* \brief Determine whether a channel ID is in the list of PJSIP channels on hold
|
|
*
|
|
* \param chan_uid - Channel being checked
|
|
*
|
|
* \retval 0 The channel is not in the hold list
|
|
* \retval 1 The channel is in the hold list
|
|
*/
|
|
static int chan_pjsip_get_hold(const char *chan_uid)
|
|
{
|
|
RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
|
|
|
|
hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
|
|
if (!hold_uid) {
|
|
return 0;
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
/*! \brief Function called to get the device state of an endpoint */
|
|
static int chan_pjsip_devicestate(const char *data)
|
|
{
|
|
RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
|
|
enum ast_device_state state = AST_DEVICE_UNKNOWN;
|
|
RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
|
|
struct ast_devstate_aggregate aggregate;
|
|
int num, inuse = 0;
|
|
|
|
if (!endpoint) {
|
|
return AST_DEVICE_INVALID;
|
|
}
|
|
|
|
endpoint_snapshot = ast_endpoint_latest_snapshot(ast_endpoint_get_tech(endpoint->persistent),
|
|
ast_endpoint_get_resource(endpoint->persistent));
|
|
|
|
if (!endpoint_snapshot) {
|
|
return AST_DEVICE_INVALID;
|
|
}
|
|
|
|
if (endpoint_snapshot->state == AST_ENDPOINT_OFFLINE) {
|
|
state = AST_DEVICE_UNAVAILABLE;
|
|
} else if (endpoint_snapshot->state == AST_ENDPOINT_ONLINE) {
|
|
state = AST_DEVICE_NOT_INUSE;
|
|
}
|
|
|
|
if (!endpoint_snapshot->num_channels) {
|
|
return state;
|
|
}
|
|
|
|
ast_devstate_aggregate_init(&aggregate);
|
|
|
|
for (num = 0; num < endpoint_snapshot->num_channels; num++) {
|
|
struct ast_channel_snapshot *snapshot;
|
|
|
|
snapshot = ast_channel_snapshot_get_latest(endpoint_snapshot->channel_ids[num]);
|
|
if (!snapshot) {
|
|
continue;
|
|
}
|
|
|
|
if (chan_pjsip_get_hold(snapshot->base->uniqueid)) {
|
|
ast_devstate_aggregate_add(&aggregate, AST_DEVICE_ONHOLD);
|
|
} else {
|
|
ast_devstate_aggregate_add(&aggregate, ast_state_chan2dev(snapshot->state));
|
|
}
|
|
|
|
if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
|
|
(snapshot->state == AST_STATE_BUSY)) {
|
|
inuse++;
|
|
}
|
|
|
|
ao2_ref(snapshot, -1);
|
|
}
|
|
|
|
if (endpoint->devicestate_busy_at && (inuse == endpoint->devicestate_busy_at)) {
|
|
state = AST_DEVICE_BUSY;
|
|
} else if (ast_devstate_aggregate_result(&aggregate) != AST_DEVICE_INVALID) {
|
|
state = ast_devstate_aggregate_result(&aggregate);
|
|
}
|
|
|
|
return state;
|
|
}
|
|
|
|
/*! \brief Function called to query options on a channel */
|
|
static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
int res = -1;
|
|
enum ast_t38_state state = T38_STATE_UNAVAILABLE;
|
|
|
|
if (!channel) {
|
|
return -1;
|
|
}
|
|
|
|
switch (option) {
|
|
case AST_OPTION_T38_STATE:
|
|
if (channel->session->endpoint->media.t38.enabled) {
|
|
switch (channel->session->t38state) {
|
|
case T38_LOCAL_REINVITE:
|
|
case T38_PEER_REINVITE:
|
|
state = T38_STATE_NEGOTIATING;
|
|
break;
|
|
case T38_ENABLED:
|
|
state = T38_STATE_NEGOTIATED;
|
|
break;
|
|
case T38_REJECTED:
|
|
state = T38_STATE_REJECTED;
|
|
break;
|
|
default:
|
|
state = T38_STATE_UNKNOWN;
|
|
break;
|
|
}
|
|
}
|
|
|
|
*((enum ast_t38_state *) data) = state;
|
|
res = 0;
|
|
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static const char *chan_pjsip_get_uniqueid(struct ast_channel *ast)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
char *uniqueid = ast_threadstorage_get(&uniqueid_threadbuf, UNIQUEID_BUFSIZE);
|
|
|
|
if (!channel || !uniqueid) {
|
|
return "";
|
|
}
|
|
|
|
ast_copy_pj_str(uniqueid, &channel->session->inv_session->dlg->call_id->id, UNIQUEID_BUFSIZE);
|
|
|
|
return uniqueid;
|
|
}
|
|
|
|
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 ((session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
|
|
(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
|
|
};
|
|
|
|
RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
|
|
struct pjsip_tx_data *tdata;
|
|
|
|
if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
|
|
session->inv_session->cause,
|
|
pjsip_get_status_text(session->inv_session->cause)->ptr);
|
|
return -1;
|
|
}
|
|
|
|
if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, 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;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief TRUE if a COLP update can be sent to the peer.
|
|
* \since 13.3.0
|
|
*
|
|
* \param session The session to see if the COLP update is allowed.
|
|
*
|
|
* \retval 0 Update is not allowed.
|
|
* \retval 1 Update is allowed.
|
|
*/
|
|
static int is_colp_update_allowed(struct ast_sip_session *session)
|
|
{
|
|
struct ast_party_id connected_id;
|
|
int update_allowed = 0;
|
|
|
|
if (!session->endpoint->id.send_connected_line
|
|
|| (!session->endpoint->id.send_pai && !session->endpoint->id.send_rpid)) {
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* Check if privacy allows the update. Check while the channel
|
|
* is locked so we can work with the shallow connected_id copy.
|
|
*/
|
|
ast_channel_lock(session->channel);
|
|
connected_id = ast_channel_connected_effective_id(session->channel);
|
|
if (connected_id.number.valid
|
|
&& (session->endpoint->id.trust_outbound
|
|
|| (ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED)) {
|
|
update_allowed = 1;
|
|
}
|
|
ast_channel_unlock(session->channel);
|
|
|
|
return update_allowed;
|
|
}
|
|
|
|
/*! \brief Update connected line information */
|
|
static int update_connected_line_information(void *data)
|
|
{
|
|
struct ast_sip_session *session = data;
|
|
|
|
if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
|
|
session->inv_session->cause,
|
|
pjsip_get_status_text(session->inv_session->cause)->ptr);
|
|
ao2_ref(session, -1);
|
|
return -1;
|
|
}
|
|
|
|
if (ast_channel_state(session->channel) == AST_STATE_UP
|
|
|| session->inv_session->role == PJSIP_ROLE_UAC) {
|
|
if (is_colp_update_allowed(session)) {
|
|
enum ast_sip_session_refresh_method method;
|
|
int generate_new_sdp;
|
|
|
|
method = session->endpoint->id.refresh_method;
|
|
if (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE) {
|
|
method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
|
|
}
|
|
|
|
/* Only the INVITE method actually needs SDP, UPDATE can do without */
|
|
generate_new_sdp = (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE);
|
|
|
|
ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp, NULL);
|
|
}
|
|
} else if (session->endpoint->id.rpid_immediate
|
|
&& session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED
|
|
&& is_colp_update_allowed(session)) {
|
|
int response_code = 0;
|
|
|
|
if (ast_channel_state(session->channel) == AST_STATE_RING) {
|
|
response_code = !session->endpoint->inband_progress ? 180 : 183;
|
|
} else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
|
|
response_code = 183;
|
|
}
|
|
|
|
if (response_code) {
|
|
struct pjsip_tx_data *packet = NULL;
|
|
|
|
if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
|
|
ast_sip_session_send_response(session, packet);
|
|
}
|
|
}
|
|
}
|
|
|
|
ao2_ref(session, -1);
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Update local hold state and send a re-INVITE with the new SDP */
|
|
static int remote_send_hold_refresh(struct ast_sip_session *session, unsigned int held)
|
|
{
|
|
struct ast_sip_session_media *session_media = session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
|
|
if (session_media) {
|
|
session_media->locally_held = held;
|
|
}
|
|
ast_sip_session_refresh(session, NULL, NULL, NULL, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, NULL);
|
|
ao2_ref(session, -1);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Update local hold state to be held */
|
|
static int remote_send_hold(void *data)
|
|
{
|
|
return remote_send_hold_refresh(data, 1);
|
|
}
|
|
|
|
/*! \brief Update local hold state to be unheld */
|
|
static int remote_send_unhold(void *data)
|
|
{
|
|
return remote_send_hold_refresh(data, 0);
|
|
}
|
|
|
|
struct topology_change_refresh_data {
|
|
struct ast_sip_session *session;
|
|
struct ast_sip_session_media_state *media_state;
|
|
};
|
|
|
|
static void topology_change_refresh_data_free(struct topology_change_refresh_data *refresh_data)
|
|
{
|
|
ao2_cleanup(refresh_data->session);
|
|
|
|
ast_sip_session_media_state_free(refresh_data->media_state);
|
|
ast_free(refresh_data);
|
|
}
|
|
|
|
static struct topology_change_refresh_data *topology_change_refresh_data_alloc(
|
|
struct ast_sip_session *session, const struct ast_stream_topology *topology)
|
|
{
|
|
struct topology_change_refresh_data *refresh_data;
|
|
|
|
refresh_data = ast_calloc(1, sizeof(*refresh_data));
|
|
if (!refresh_data) {
|
|
return NULL;
|
|
}
|
|
|
|
refresh_data->session = ao2_bump(session);
|
|
refresh_data->media_state = ast_sip_session_media_state_alloc();
|
|
if (!refresh_data->media_state) {
|
|
topology_change_refresh_data_free(refresh_data);
|
|
return NULL;
|
|
}
|
|
refresh_data->media_state->topology = ast_stream_topology_clone(topology);
|
|
if (!refresh_data->media_state->topology) {
|
|
topology_change_refresh_data_free(refresh_data);
|
|
return NULL;
|
|
}
|
|
|
|
return refresh_data;
|
|
}
|
|
|
|
static int on_topology_change_response(struct ast_sip_session *session, pjsip_rx_data *rdata)
|
|
{
|
|
SCOPE_ENTER(3, "%s: Received response code %d. PT: %s AT: %s\n", ast_sip_session_get_name(session),
|
|
rdata->msg_info.msg->line.status.code,
|
|
ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
|
|
|
|
|
|
if (PJSIP_IS_STATUS_IN_CLASS(rdata->msg_info.msg->line.status.code, 200)) {
|
|
/* The topology was changed to something new so give notice to what requested
|
|
* it so it queries the channel and updates accordingly.
|
|
*/
|
|
if (session->channel) {
|
|
ast_queue_control(session->channel, AST_CONTROL_STREAM_TOPOLOGY_CHANGED);
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s: Queued topology change frame\n", ast_sip_session_get_name(session));
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s: No channel? Can't queue topology change frame\n", ast_sip_session_get_name(session));
|
|
} else if (300 <= rdata->msg_info.msg->line.status.code) {
|
|
/* The topology change failed, so drop the current pending media state */
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s: response code > 300. Resetting pending media state\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s: Nothing to do\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
static int send_topology_change_refresh(void *data)
|
|
{
|
|
struct topology_change_refresh_data *refresh_data = data;
|
|
struct ast_sip_session *session = refresh_data->session;
|
|
enum ast_channel_state state = ast_channel_state(session->channel);
|
|
enum ast_sip_session_refresh_method method = AST_SIP_SESSION_REFRESH_METHOD_INVITE;
|
|
int ret;
|
|
SCOPE_ENTER(3, "%s: %s\n", ast_sip_session_get_name(session),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(refresh_data->media_state->topology, &STR_TMP)));
|
|
|
|
/* See RFC 6337, especially section 3.2: If the early media SDP was sent reliably, we are allowed
|
|
* to send UPDATEs. Only relevant for AST_STATE_RINGING and AST_STATE_RING - if the channel is UP,
|
|
* re-INVITES can be sent.
|
|
*/
|
|
if (session->early_confirmed && (state == AST_STATE_RINGING || state == AST_STATE_RING)) {
|
|
method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
|
|
}
|
|
|
|
ret = ast_sip_session_refresh(session, NULL, NULL, on_topology_change_response,
|
|
method, 1, refresh_data->media_state);
|
|
refresh_data->media_state = NULL;
|
|
topology_change_refresh_data_free(refresh_data);
|
|
|
|
SCOPE_EXIT_RTN_VALUE(ret, "%s\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
static int handle_topology_request_change(struct ast_sip_session *session,
|
|
const struct ast_stream_topology *proposed)
|
|
{
|
|
struct topology_change_refresh_data *refresh_data;
|
|
int res;
|
|
SCOPE_ENTER(1);
|
|
|
|
refresh_data = topology_change_refresh_data_alloc(session, proposed);
|
|
if (!refresh_data) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create refresh_data\n");
|
|
}
|
|
|
|
res = ast_sip_push_task(session->serializer, send_topology_change_refresh, refresh_data);
|
|
if (res) {
|
|
topology_change_refresh_data_free(refresh_data);
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
|
|
}
|
|
|
|
/* Forward declarations */
|
|
static int transmit_info_dtmf(void *data);
|
|
static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration);
|
|
|
|
/*! \brief Function called by core to ask the channel to indicate some sort of condition */
|
|
static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session_media *media;
|
|
int response_code = 0;
|
|
int res = 0;
|
|
char *device_buf;
|
|
size_t device_buf_size;
|
|
int i;
|
|
const struct ast_stream_topology *topology;
|
|
struct ast_frame f = {
|
|
.frametype = AST_FRAME_CONTROL,
|
|
.subclass = {
|
|
.integer = condition
|
|
},
|
|
.datalen = datalen,
|
|
.data.ptr = (void *)data,
|
|
};
|
|
char condition_name[256];
|
|
unsigned int duration;
|
|
char digit;
|
|
struct info_dtmf_data *dtmf_data;
|
|
|
|
SCOPE_ENTER(3, "%s: Indicated %s\n", ast_channel_name(ast),
|
|
ast_frame_subclass2str(&f, condition_name, sizeof(condition_name), NULL, 0));
|
|
|
|
switch (condition) {
|
|
case AST_CONTROL_RINGING:
|
|
if (ast_channel_state(ast) == AST_STATE_RING) {
|
|
if (channel->session->endpoint->inband_progress ||
|
|
(channel->session->inv_session && channel->session->inv_session->neg &&
|
|
pjmedia_sdp_neg_get_state(channel->session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE)) {
|
|
res = -1;
|
|
if (ast_sip_get_allow_sending_180_after_183()) {
|
|
response_code = 180;
|
|
} else {
|
|
response_code = 183;
|
|
}
|
|
} else {
|
|
response_code = 180;
|
|
}
|
|
} else {
|
|
res = -1;
|
|
}
|
|
ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
|
|
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;
|
|
}
|
|
ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
|
|
break;
|
|
case AST_CONTROL_FLASH:
|
|
duration = 300;
|
|
digit = '!';
|
|
dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
|
|
|
|
if (!dtmf_data) {
|
|
res = -1;
|
|
break;
|
|
}
|
|
|
|
if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
|
|
ast_log(LOG_WARNING, "Error sending FLASH via INFO on channel %s\n", ast_channel_name(ast));
|
|
ao2_ref(dtmf_data, -1); /* dtmf_data can't be null here */
|
|
res = -1;
|
|
}
|
|
break;
|
|
case AST_CONTROL_VIDUPDATE:
|
|
for (i = 0; i < AST_VECTOR_SIZE(&channel->session->active_media_state->sessions); ++i) {
|
|
media = AST_VECTOR_GET(&channel->session->active_media_state->sessions, i);
|
|
if (!media || media->type != AST_MEDIA_TYPE_VIDEO) {
|
|
continue;
|
|
}
|
|
if (media->rtp) {
|
|
/* FIXME: Only use this for VP8. Additional work would have to be done to
|
|
* fully support other video codecs */
|
|
|
|
if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp8) != AST_FORMAT_CMP_NOT_EQUAL ||
|
|
ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_vp9) != AST_FORMAT_CMP_NOT_EQUAL ||
|
|
ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h265) != AST_FORMAT_CMP_NOT_EQUAL ||
|
|
(channel->session->endpoint->media.webrtc &&
|
|
ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), ast_format_h264) != AST_FORMAT_CMP_NOT_EQUAL)) {
|
|
/* FIXME Fake RTP write, this will be sent as an RTCP packet. Ideally the
|
|
* RTP engine would provide a way to externally write/schedule RTCP
|
|
* packets */
|
|
struct ast_frame fr;
|
|
fr.frametype = AST_FRAME_CONTROL;
|
|
fr.subclass.integer = AST_CONTROL_VIDUPDATE;
|
|
res = ast_rtp_instance_write(media->rtp, &fr);
|
|
} else {
|
|
ao2_ref(channel->session, +1);
|
|
if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
|
|
ao2_cleanup(channel->session);
|
|
}
|
|
}
|
|
ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Success");
|
|
} else {
|
|
ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Failure");
|
|
res = -1;
|
|
}
|
|
}
|
|
/* XXX If there were no video streams, then this should set
|
|
* res to -1
|
|
*/
|
|
break;
|
|
case AST_CONTROL_CONNECTED_LINE:
|
|
ao2_ref(channel->session, +1);
|
|
if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
|
|
ao2_cleanup(channel->session);
|
|
}
|
|
break;
|
|
case AST_CONTROL_UPDATE_RTP_PEER:
|
|
break;
|
|
case AST_CONTROL_PVT_CAUSE_CODE:
|
|
res = -1;
|
|
break;
|
|
case AST_CONTROL_MASQUERADE_NOTIFY:
|
|
ast_assert(datalen == sizeof(int));
|
|
if (*(int *) data) {
|
|
/*
|
|
* Masquerade is beginning:
|
|
* Wait for session serializer to get suspended.
|
|
*/
|
|
ast_channel_unlock(ast);
|
|
ast_sip_session_suspend(channel->session);
|
|
ast_channel_lock(ast);
|
|
} else {
|
|
/*
|
|
* Masquerade is complete:
|
|
* Unsuspend the session serializer.
|
|
*/
|
|
ast_sip_session_unsuspend(channel->session);
|
|
}
|
|
break;
|
|
case AST_CONTROL_HOLD:
|
|
chan_pjsip_add_hold(ast_channel_uniqueid(ast));
|
|
device_buf_size = strlen(ast_channel_name(ast)) + 1;
|
|
device_buf = alloca(device_buf_size);
|
|
ast_channel_get_device_name(ast, device_buf, device_buf_size);
|
|
ast_devstate_changed_literal(AST_DEVICE_ONHOLD, 1, device_buf);
|
|
if (!channel->session->moh_passthrough) {
|
|
ast_moh_start(ast, data, NULL);
|
|
} else {
|
|
if (ast_sip_push_task(channel->session->serializer, remote_send_hold, ao2_bump(channel->session))) {
|
|
ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n",
|
|
ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
|
|
ao2_ref(channel->session, -1);
|
|
}
|
|
}
|
|
break;
|
|
case AST_CONTROL_UNHOLD:
|
|
chan_pjsip_remove_hold(ast_channel_uniqueid(ast));
|
|
device_buf_size = strlen(ast_channel_name(ast)) + 1;
|
|
device_buf = alloca(device_buf_size);
|
|
ast_channel_get_device_name(ast, device_buf, device_buf_size);
|
|
ast_devstate_changed_literal(AST_DEVICE_UNKNOWN, 1, device_buf);
|
|
if (!channel->session->moh_passthrough) {
|
|
ast_moh_stop(ast);
|
|
} else {
|
|
if (ast_sip_push_task(channel->session->serializer, remote_send_unhold, ao2_bump(channel->session))) {
|
|
ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n",
|
|
ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
|
|
ao2_ref(channel->session, -1);
|
|
}
|
|
}
|
|
break;
|
|
case AST_CONTROL_SRCUPDATE:
|
|
break;
|
|
case AST_CONTROL_SRCCHANGE:
|
|
break;
|
|
case AST_CONTROL_REDIRECTING:
|
|
if (ast_channel_state(ast) != AST_STATE_UP) {
|
|
response_code = 181;
|
|
} else {
|
|
res = -1;
|
|
}
|
|
break;
|
|
case AST_CONTROL_T38_PARAMETERS:
|
|
res = 0;
|
|
|
|
if (channel->session->t38state == T38_PEER_REINVITE) {
|
|
const struct ast_control_t38_parameters *parameters = data;
|
|
|
|
if (parameters->request_response == AST_T38_REQUEST_PARMS) {
|
|
res = AST_T38_REQUEST_PARMS;
|
|
}
|
|
}
|
|
|
|
break;
|
|
case AST_CONTROL_STREAM_TOPOLOGY_REQUEST_CHANGE:
|
|
topology = data;
|
|
ast_trace(-1, "%s: New topology: %s\n", ast_channel_name(ast),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
|
|
res = handle_topology_request_change(channel->session, topology);
|
|
break;
|
|
case AST_CONTROL_STREAM_TOPOLOGY_CHANGED:
|
|
break;
|
|
case AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED:
|
|
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 (response_code) {
|
|
struct indicate_data *ind_data = indicate_data_alloc(channel->session, condition, response_code, data, datalen);
|
|
|
|
if (!ind_data) {
|
|
SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc indicate data\n", ast_channel_name(ast));
|
|
}
|
|
|
|
if (ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
|
|
ast_log(LOG_ERROR, "%s: Cannot send response code %d to endpoint %s. Could not queue task properly\n",
|
|
ast_channel_name(ast), response_code, ast_sorcery_object_get_id(channel->session->endpoint));
|
|
ao2_cleanup(ind_data);
|
|
res = -1;
|
|
}
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(res, "%s\n", ast_channel_name(ast));
|
|
}
|
|
|
|
struct transfer_data {
|
|
struct ast_sip_session *session;
|
|
char *target;
|
|
};
|
|
|
|
static void transfer_data_destroy(void *obj)
|
|
{
|
|
struct transfer_data *trnf_data = obj;
|
|
|
|
ast_free(trnf_data->target);
|
|
ao2_cleanup(trnf_data->session);
|
|
}
|
|
|
|
static struct transfer_data *transfer_data_alloc(struct ast_sip_session *session, const char *target)
|
|
{
|
|
struct transfer_data *trnf_data = ao2_alloc(sizeof(*trnf_data), transfer_data_destroy);
|
|
|
|
if (!trnf_data) {
|
|
return NULL;
|
|
}
|
|
|
|
if (!(trnf_data->target = ast_strdup(target))) {
|
|
ao2_ref(trnf_data, -1);
|
|
return NULL;
|
|
}
|
|
|
|
ao2_ref(session, +1);
|
|
trnf_data->session = session;
|
|
|
|
return trnf_data;
|
|
}
|
|
|
|
static void transfer_redirect(struct ast_sip_session *session, const char *target)
|
|
{
|
|
pjsip_tx_data *packet;
|
|
enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
|
|
pjsip_contact_hdr *contact;
|
|
pj_str_t tmp;
|
|
|
|
if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS
|
|
|| !packet) {
|
|
ast_log(LOG_WARNING, "Failed to redirect PJSIP session for channel %s\n",
|
|
ast_channel_name(session->channel));
|
|
message = AST_TRANSFER_FAILED;
|
|
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
|
|
return;
|
|
}
|
|
|
|
if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
|
|
contact = pjsip_contact_hdr_create(packet->pool);
|
|
}
|
|
|
|
pj_strdup2_with_null(packet->pool, &tmp, target);
|
|
if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
|
|
ast_log(LOG_WARNING, "Failed to parse destination URI '%s' for channel %s\n",
|
|
target, ast_channel_name(session->channel));
|
|
message = AST_TRANSFER_FAILED;
|
|
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
pjsip_tx_data_dec_ref(packet);
|
|
|
|
return;
|
|
}
|
|
pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
|
|
|
|
ast_sip_session_send_response(session, packet);
|
|
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
}
|
|
|
|
/*! \brief REFER Callback module, used to attach session data structure to subscription */
|
|
static pjsip_module refer_callback_module = {
|
|
.name = { "REFER Callback", 14 },
|
|
.id = -1,
|
|
};
|
|
|
|
/*!
|
|
* \brief Callback function to report status of implicit REFER-NOTIFY subscription.
|
|
*
|
|
* This function will be called on any state change in the REFER-NOTIFY subscription.
|
|
* Its primary purpose is to report SUCCESS/FAILURE of a transfer initiated via
|
|
* \ref transfer_refer as well as to terminate the subscription, if necessary.
|
|
*/
|
|
static void xfer_client_on_evsub_state(pjsip_evsub *sub, pjsip_event *event)
|
|
{
|
|
struct ast_channel *chan;
|
|
enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
|
|
int res = 0;
|
|
|
|
if (!event) {
|
|
return;
|
|
}
|
|
|
|
chan = pjsip_evsub_get_mod_data(sub, refer_callback_module.id);
|
|
if (!chan) {
|
|
return;
|
|
}
|
|
|
|
if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
|
|
/* Check if subscription is suppressed and terminate and send completion code, if so. */
|
|
pjsip_rx_data *rdata;
|
|
pjsip_generic_string_hdr *refer_sub;
|
|
const pj_str_t REFER_SUB = { "Refer-Sub", 9 };
|
|
|
|
ast_debug(3, "Transfer accepted on channel %s\n", ast_channel_name(chan));
|
|
|
|
/* Check if response message */
|
|
if (event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
|
|
rdata = event->body.tsx_state.src.rdata;
|
|
|
|
/* Find Refer-Sub header */
|
|
refer_sub = pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &REFER_SUB, NULL);
|
|
|
|
/* Check if subscription is suppressed. If it is, the far end will not terminate it,
|
|
* and the subscription will remain active until it times out. Terminating it here
|
|
* eliminates the unnecessary timeout.
|
|
*/
|
|
if (refer_sub && !pj_stricmp2(&refer_sub->hvalue, "false")) {
|
|
/* Since no subscription is desired, assume that call has been transferred successfully. */
|
|
/* Channel reference will be released at end of function */
|
|
/* Terminate subscription. */
|
|
pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
|
|
pjsip_evsub_terminate(sub, PJ_TRUE);
|
|
res = -1;
|
|
}
|
|
}
|
|
} else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
|
|
pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
|
|
/* Check for NOTIFY complete or error. */
|
|
pjsip_msg *msg;
|
|
pjsip_msg_body *body;
|
|
pjsip_status_line status_line = { .code = 0 };
|
|
pj_bool_t is_last;
|
|
pj_status_t status;
|
|
|
|
if (event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
|
|
pjsip_rx_data *rdata;
|
|
|
|
rdata = event->body.tsx_state.src.rdata;
|
|
msg = rdata->msg_info.msg;
|
|
|
|
if (msg->type == PJSIP_REQUEST_MSG) {
|
|
if (!pjsip_method_cmp(&msg->line.req.method, pjsip_get_notify_method())) {
|
|
body = msg->body;
|
|
if (body && !pj_stricmp2(&body->content_type.type, "message")
|
|
&& !pj_stricmp2(&body->content_type.subtype, "sipfrag")) {
|
|
pjsip_parse_status_line((char *)body->data, body->len, &status_line);
|
|
}
|
|
}
|
|
} else {
|
|
status_line.code = msg->line.status.code;
|
|
status_line.reason = msg->line.status.reason;
|
|
}
|
|
} else {
|
|
status_line.code = 500;
|
|
status_line.reason = *pjsip_get_status_text(500);
|
|
}
|
|
|
|
is_last = (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED);
|
|
/* If the status code is >= 200, the subscription is finished. */
|
|
if (status_line.code >= 200 || is_last) {
|
|
res = -1;
|
|
|
|
/* If the subscription has terminated, return AST_TRANSFER_SUCCESS for 2XX.
|
|
* Return AST_TRANSFER_FAILED for any code < 200.
|
|
* Otherwise, return the status code.
|
|
* The subscription should not terminate for any code < 200,
|
|
* but if it does, that constitutes a failure. */
|
|
if (status_line.code < 200) {
|
|
message = AST_TRANSFER_FAILED;
|
|
} else if (status_line.code >= 300) {
|
|
message = status_line.code;
|
|
}
|
|
|
|
/* If subscription not terminated and subscription is finished (status code >= 200)
|
|
* terminate it */
|
|
if (!is_last) {
|
|
pjsip_tx_data *tdata;
|
|
|
|
status = pjsip_evsub_initiate(sub, pjsip_get_subscribe_method(), 0, &tdata);
|
|
if (status == PJ_SUCCESS) {
|
|
pjsip_evsub_send_request(sub, tdata);
|
|
}
|
|
}
|
|
/* Finished. Remove session from subscription */
|
|
pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
|
|
ast_debug(3, "Transfer channel %s completed: %d %.*s (%s)\n",
|
|
ast_channel_name(chan),
|
|
status_line.code,
|
|
(int)status_line.reason.slen, status_line.reason.ptr,
|
|
(message == AST_TRANSFER_SUCCESS) ? "Success" : "Failure");
|
|
}
|
|
}
|
|
|
|
if (res) {
|
|
ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
ao2_ref(chan, -1);
|
|
}
|
|
}
|
|
|
|
static void transfer_refer(struct ast_sip_session *session, const char *target)
|
|
{
|
|
pjsip_evsub *sub;
|
|
enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
|
|
pj_str_t tmp;
|
|
pjsip_tx_data *packet;
|
|
const char *ref_by_val;
|
|
char local_info[pj_strlen(&session->inv_session->dlg->local.info_str) + 1];
|
|
struct pjsip_evsub_user xfer_cb;
|
|
struct ast_channel *chan = session->channel;
|
|
|
|
pj_bzero(&xfer_cb, sizeof(xfer_cb));
|
|
xfer_cb.on_evsub_state = &xfer_client_on_evsub_state;
|
|
|
|
if (pjsip_xfer_create_uac(session->inv_session->dlg, &xfer_cb, &sub) != PJ_SUCCESS) {
|
|
message = AST_TRANSFER_FAILED;
|
|
ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
|
|
return;
|
|
}
|
|
|
|
/* refer_callback_module requires a reference to chan
|
|
* which will be released in xfer_client_on_evsub_state()
|
|
* when the implicit REFER subscription terminates */
|
|
pjsip_evsub_set_mod_data(sub, refer_callback_module.id, chan);
|
|
ao2_ref(chan, +1);
|
|
|
|
if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
|
|
goto failure;
|
|
}
|
|
|
|
ref_by_val = pbx_builtin_getvar_helper(chan, "SIPREFERREDBYHDR");
|
|
if (!ast_strlen_zero(ref_by_val)) {
|
|
ast_sip_add_header(packet, "Referred-By", ref_by_val);
|
|
} else {
|
|
ast_copy_pj_str(local_info, &session->inv_session->dlg->local.info_str, sizeof(local_info));
|
|
ast_sip_add_header(packet, "Referred-By", local_info);
|
|
}
|
|
|
|
if (pjsip_xfer_send_request(sub, packet) == PJ_SUCCESS) {
|
|
return;
|
|
}
|
|
|
|
failure:
|
|
message = AST_TRANSFER_FAILED;
|
|
ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
|
|
pjsip_evsub_terminate(sub, PJ_FALSE);
|
|
|
|
ao2_ref(chan, -1);
|
|
}
|
|
|
|
static int transfer(void *data)
|
|
{
|
|
struct transfer_data *trnf_data = data;
|
|
struct ast_sip_endpoint *endpoint = NULL;
|
|
struct ast_sip_contact *contact = NULL;
|
|
const char *target = trnf_data->target;
|
|
|
|
if (trnf_data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
|
|
trnf_data->session->inv_session->cause,
|
|
pjsip_get_status_text(trnf_data->session->inv_session->cause)->ptr);
|
|
} else {
|
|
/* See if we have an endpoint; if so, use its contact */
|
|
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", target);
|
|
if (endpoint) {
|
|
contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
|
|
if (contact && !ast_strlen_zero(contact->uri)) {
|
|
target = contact->uri;
|
|
}
|
|
}
|
|
|
|
if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
|
|
transfer_redirect(trnf_data->session, target);
|
|
} else {
|
|
transfer_refer(trnf_data->session, target);
|
|
}
|
|
}
|
|
|
|
ao2_ref(trnf_data, -1);
|
|
ao2_cleanup(endpoint);
|
|
ao2_cleanup(contact);
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core for Asterisk initiated transfer */
|
|
static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
|
|
struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
|
|
|
|
if (!trnf_data) {
|
|
return -1;
|
|
}
|
|
|
|
if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
|
|
ast_log(LOG_WARNING, "Error requesting transfer\n");
|
|
ao2_cleanup(trnf_data);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to start a DTMF digit */
|
|
static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
|
|
struct ast_sip_session_media *media;
|
|
|
|
media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
|
|
|
|
switch (channel->session->dtmf) {
|
|
case AST_SIP_DTMF_RFC_4733:
|
|
if (!media || !media->rtp) {
|
|
return 0;
|
|
}
|
|
|
|
ast_rtp_instance_dtmf_begin(media->rtp, digit);
|
|
break;
|
|
case AST_SIP_DTMF_AUTO:
|
|
if (!media || !media->rtp) {
|
|
return 0;
|
|
}
|
|
|
|
if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
|
|
return -1;
|
|
}
|
|
|
|
ast_rtp_instance_dtmf_begin(media->rtp, digit);
|
|
break;
|
|
case AST_SIP_DTMF_AUTO_INFO:
|
|
if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_NONE)) {
|
|
return 0;
|
|
}
|
|
ast_rtp_instance_dtmf_begin(media->rtp, digit);
|
|
break;
|
|
case AST_SIP_DTMF_NONE:
|
|
break;
|
|
case AST_SIP_DTMF_INBAND:
|
|
return -1;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
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 (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
|
|
session->inv_session->cause,
|
|
pjsip_get_status_text(session->inv_session->cause)->ptr);
|
|
return -1;
|
|
}
|
|
|
|
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, 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 chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct ast_sip_session_media *media;
|
|
|
|
if (!channel || !channel->session) {
|
|
/* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */
|
|
ast_debug(3, "Channel %s disappeared while calling digit_end\n", ast_channel_name(ast));
|
|
return -1;
|
|
}
|
|
|
|
media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
|
|
|
|
switch (channel->session->dtmf) {
|
|
case AST_SIP_DTMF_AUTO_INFO:
|
|
{
|
|
if (!media || !media->rtp) {
|
|
return 0;
|
|
}
|
|
|
|
if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_NONE) {
|
|
ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
|
|
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
|
|
break;
|
|
}
|
|
/* If RFC_4733 was not negotiated, fail through to the DTMF_INFO processing */
|
|
ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 NOT negotiated using INFO instead.\n", ast_channel_name(ast));
|
|
}
|
|
|
|
case AST_SIP_DTMF_INFO:
|
|
{
|
|
struct info_dtmf_data *dtmf_data = info_dtmf_data_alloc(channel->session, digit, duration);
|
|
|
|
if (!dtmf_data) {
|
|
return -1;
|
|
}
|
|
|
|
if (ast_sip_push_task(channel->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 0;
|
|
}
|
|
|
|
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
|
|
break;
|
|
case AST_SIP_DTMF_AUTO:
|
|
if (!media || !media->rtp) {
|
|
return 0;
|
|
}
|
|
|
|
if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
|
|
return -1;
|
|
}
|
|
|
|
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
|
|
break;
|
|
case AST_SIP_DTMF_NONE:
|
|
break;
|
|
case AST_SIP_DTMF_INBAND:
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void update_initial_connected_line(struct ast_sip_session *session)
|
|
{
|
|
struct ast_party_connected_line connected;
|
|
|
|
/*
|
|
* Use the channel CALLERID() as the initial connected line data.
|
|
* The core or a predial handler may have supplied missing values
|
|
* from the session->endpoint->id.self about who we are calling.
|
|
*/
|
|
ast_channel_lock(session->channel);
|
|
ast_party_id_copy(&session->id, &ast_channel_caller(session->channel)->id);
|
|
ast_channel_unlock(session->channel);
|
|
|
|
/* Supply initial connected line information if available. */
|
|
if (!session->id.number.valid && !session->id.name.valid) {
|
|
return;
|
|
}
|
|
|
|
ast_party_connected_line_init(&connected);
|
|
connected.id = session->id;
|
|
connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
|
|
|
|
ast_channel_queue_connected_line_update(session->channel, &connected, NULL);
|
|
}
|
|
|
|
static int call(void *data)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = data;
|
|
struct ast_sip_session *session = channel->session;
|
|
pjsip_tx_data *tdata;
|
|
int res = 0;
|
|
SCOPE_ENTER(1, "%s Topology: %s\n",
|
|
ast_sip_session_get_name(session),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(channel->session->pending_media_state->topology, &STR_TMP))
|
|
);
|
|
|
|
|
|
res = ast_sip_session_create_invite(session, &tdata);
|
|
|
|
if (res) {
|
|
ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
|
|
ast_queue_hangup(session->channel);
|
|
} else {
|
|
set_channel_on_rtp_instance(session, ast_channel_uniqueid(session->channel));
|
|
update_initial_connected_line(session);
|
|
ast_sip_session_send_request(session, tdata);
|
|
}
|
|
ao2_ref(channel, -1);
|
|
SCOPE_EXIT_RTN_VALUE(res, "RC: %d\n", res);
|
|
}
|
|
|
|
/*! \brief Function called by core to actually start calling a remote party */
|
|
static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
SCOPE_ENTER(1, "%s Topology: %s\n", ast_sip_session_get_name(channel->session),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(channel->session->pending_media_state->topology, &STR_TMP)));
|
|
|
|
ao2_ref(channel, +1);
|
|
if (ast_sip_push_task(channel->session->serializer, call, channel)) {
|
|
ast_log(LOG_WARNING, "Error attempting to place outbound call to '%s'\n", dest);
|
|
ao2_cleanup(channel);
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't push task\n");
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0, "'call' task pushed\n");
|
|
}
|
|
|
|
/*! \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;
|
|
}
|
|
|
|
/*! \brief Clear a channel from a session along with its PVT */
|
|
static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast)
|
|
{
|
|
session->channel = NULL;
|
|
set_channel_on_rtp_instance(session, "");
|
|
ast_channel_tech_pvt_set(ast, NULL);
|
|
}
|
|
|
|
static int hangup(void *data)
|
|
{
|
|
struct hangup_data *h_data = data;
|
|
struct ast_channel *ast = h_data->chan;
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
|
|
|
|
/*
|
|
* Before cleaning we have to ensure that channel or its session is not NULL
|
|
* we have seen rare case when taskprocessor calls hangup but channel is NULL
|
|
* due to SIP session timeout and answer happening at the same time
|
|
*/
|
|
if (channel) {
|
|
struct ast_sip_session *session = channel->session;
|
|
if (session) {
|
|
int cause = h_data->cause;
|
|
|
|
if (channel->session->active_media_state &&
|
|
channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
|
|
struct ast_sip_session_media *media =
|
|
channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
|
|
if (media->rtp) {
|
|
ast_rtp_instance_set_stats_vars(ast, media->rtp);
|
|
}
|
|
}
|
|
|
|
/*
|
|
* It's possible that session_terminate might cause the session to be destroyed
|
|
* immediately so we need to keep a reference to it so we can NULL session->channel
|
|
* afterwards.
|
|
*/
|
|
ast_sip_session_terminate(ao2_bump(session), cause);
|
|
clear_session_and_channel(session, ast);
|
|
ao2_cleanup(session);
|
|
}
|
|
ao2_cleanup(channel);
|
|
}
|
|
ao2_cleanup(h_data);
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
/*! \brief Function called by core to hang up a PJSIP session */
|
|
static int chan_pjsip_hangup(struct ast_channel *ast)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
int cause;
|
|
struct hangup_data *h_data;
|
|
SCOPE_ENTER(1, "%s\n", ast_channel_name(ast));
|
|
|
|
if (!channel || !channel->session) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "No channel or session\n");
|
|
}
|
|
|
|
cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
|
|
h_data = hangup_data_alloc(cause, ast);
|
|
|
|
if (!h_data) {
|
|
goto failure;
|
|
}
|
|
|
|
if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
|
|
ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
|
|
goto failure;
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0, "Cause: %d\n", cause);
|
|
|
|
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
|
|
*/
|
|
clear_session_and_channel(channel->session, ast);
|
|
ao2_cleanup(channel);
|
|
ao2_cleanup(h_data);
|
|
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Cause: %d\n", cause);
|
|
}
|
|
|
|
struct request_data {
|
|
struct ast_sip_session *session;
|
|
struct ast_stream_topology *topology;
|
|
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;
|
|
struct ast_sip_endpoint *endpoint;
|
|
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(endpoint);
|
|
AST_APP_ARG(aor);
|
|
);
|
|
SCOPE_ENTER(1, "%s\n",tmp);
|
|
|
|
if (ast_strlen_zero(tmp)) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
|
|
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Empty destination\n");
|
|
}
|
|
|
|
AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
|
|
|
|
if (ast_sip_get_disable_multi_domain()) {
|
|
/* 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)) {
|
|
if (request_user) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@<endpoint-name>\n",
|
|
request_user);
|
|
} else {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
|
|
}
|
|
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
|
|
}
|
|
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
|
|
endpoint_name);
|
|
if (!endpoint) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
|
|
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
|
|
}
|
|
} else {
|
|
/* First try to find an exact endpoint match, for single (user) or multi-domain (user@domain) */
|
|
endpoint_name = args.endpoint;
|
|
if (ast_strlen_zero(endpoint_name)) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
|
|
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
|
|
}
|
|
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
|
|
endpoint_name);
|
|
if (!endpoint) {
|
|
/* It seems it's not a multi-domain endpoint or single endpoint exact match,
|
|
* it's possible that it's a SIP trunk with a specified user (user@trunkname),
|
|
* so extract the user before @ sign.
|
|
*/
|
|
endpoint_name = strchr(args.endpoint, '@');
|
|
if (!endpoint_name) {
|
|
/*
|
|
* Couldn't find an '@' so it had to be an endpoint
|
|
* name that doesn't exist.
|
|
*/
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n",
|
|
args.endpoint);
|
|
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
|
|
}
|
|
request_user = args.endpoint;
|
|
*endpoint_name++ = '\0';
|
|
|
|
if (ast_strlen_zero(endpoint_name)) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name: %s@<endpoint-name>\n",
|
|
request_user);
|
|
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Empty endpoint name\n");
|
|
}
|
|
|
|
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint",
|
|
endpoint_name);
|
|
if (!endpoint) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
|
|
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Endpoint not found\n");
|
|
}
|
|
}
|
|
}
|
|
|
|
session = ast_sip_session_create_outgoing(endpoint, NULL, args.aor, request_user,
|
|
req_data->topology);
|
|
ao2_ref(endpoint, -1);
|
|
if (!session) {
|
|
ast_log(LOG_ERROR, "Failed to create outgoing session to endpoint '%s'\n", endpoint_name);
|
|
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create session\n");
|
|
}
|
|
|
|
req_data->session = session;
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
/*! \brief Function called by core to create a new outgoing PJSIP session */
|
|
static struct ast_channel *chan_pjsip_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
|
|
{
|
|
struct request_data req_data;
|
|
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
|
|
SCOPE_ENTER(1, "%s Topology: %s\n", data,
|
|
ast_str_tmp(256, ast_stream_topology_to_str(topology, &STR_TMP)));
|
|
|
|
req_data.topology = topology;
|
|
req_data.dest = data;
|
|
/* Default failure value in case ast_sip_push_task_wait_servant() itself fails. */
|
|
req_data.cause = AST_CAUSE_FAILURE;
|
|
|
|
if (ast_sip_push_task_wait_servant(NULL, request, &req_data)) {
|
|
*cause = req_data.cause;
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't push task\n");
|
|
}
|
|
|
|
session = req_data.session;
|
|
|
|
if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) {
|
|
/* Session needs to be terminated prematurely */
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create channel\n");
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(session->channel, "Channel: %s\n", ast_channel_name(session->channel));
|
|
}
|
|
|
|
static struct ast_channel *chan_pjsip_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)
|
|
{
|
|
struct ast_stream_topology *topology;
|
|
struct ast_channel *chan;
|
|
|
|
topology = ast_stream_topology_create_from_format_cap(cap);
|
|
if (!topology) {
|
|
return NULL;
|
|
}
|
|
|
|
chan = chan_pjsip_request_with_stream_topology(type, topology, assignedids, requestor, data, cause);
|
|
|
|
ast_stream_topology_free(topology);
|
|
|
|
return chan;
|
|
}
|
|
|
|
struct sendtext_data {
|
|
struct ast_sip_session *session;
|
|
struct ast_msg_data *msg;
|
|
};
|
|
|
|
static void sendtext_data_destroy(void *obj)
|
|
{
|
|
struct sendtext_data *data = obj;
|
|
ao2_cleanup(data->session);
|
|
ast_free(data->msg);
|
|
}
|
|
|
|
static struct sendtext_data* sendtext_data_create(struct ast_channel *chan,
|
|
struct ast_msg_data *msg)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
|
|
struct sendtext_data *data = ao2_alloc(sizeof(*data), sendtext_data_destroy);
|
|
|
|
if (!data) {
|
|
return NULL;
|
|
}
|
|
|
|
data->msg = ast_msg_data_dup(msg);
|
|
if (!data->msg) {
|
|
ao2_cleanup(data);
|
|
return NULL;
|
|
}
|
|
data->session = channel->session;
|
|
ao2_ref(data->session, +1);
|
|
|
|
return data;
|
|
}
|
|
|
|
static int sendtext(void *obj)
|
|
{
|
|
struct sendtext_data *data = obj;
|
|
pjsip_tx_data *tdata;
|
|
const char *body_text = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_BODY);
|
|
const char *content_type = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_CONTENT_TYPE);
|
|
char *sep;
|
|
struct ast_sip_body body = {
|
|
.type = "text",
|
|
.subtype = "plain",
|
|
.body_text = body_text,
|
|
};
|
|
|
|
if (!ast_strlen_zero(content_type)) {
|
|
sep = strchr(content_type, '/');
|
|
if (sep) {
|
|
*sep = '\0';
|
|
body.type = content_type;
|
|
body.subtype = ++sep;
|
|
}
|
|
}
|
|
|
|
if (data->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
|
|
data->session->inv_session->cause,
|
|
pjsip_get_status_text(data->session->inv_session->cause)->ptr);
|
|
} else {
|
|
pjsip_from_hdr *hdr;
|
|
pjsip_name_addr *name_addr;
|
|
const char *from = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_FROM);
|
|
const char *to = ast_msg_data_get_attribute(data->msg, AST_MSG_DATA_ATTR_TO);
|
|
int invalidate_tdata = 0;
|
|
|
|
ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, NULL, &tdata);
|
|
ast_sip_add_body(tdata, &body);
|
|
|
|
/*
|
|
* If we have a 'from' in the msg, set the display name in the From
|
|
* header to it.
|
|
*/
|
|
if (!ast_strlen_zero(from)) {
|
|
hdr = PJSIP_MSG_FROM_HDR(tdata->msg);
|
|
name_addr = (pjsip_name_addr *) hdr->uri;
|
|
pj_strdup2(tdata->pool, &name_addr->display, from);
|
|
invalidate_tdata = 1;
|
|
}
|
|
|
|
/*
|
|
* If we have a 'to' in the msg, set the display name in the To
|
|
* header to it.
|
|
*/
|
|
if (!ast_strlen_zero(to)) {
|
|
hdr = PJSIP_MSG_TO_HDR(tdata->msg);
|
|
name_addr = (pjsip_name_addr *) hdr->uri;
|
|
pj_strdup2(tdata->pool, &name_addr->display, to);
|
|
invalidate_tdata = 1;
|
|
}
|
|
|
|
if (invalidate_tdata) {
|
|
pjsip_tx_data_invalidate_msg(tdata);
|
|
}
|
|
|
|
ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint, NULL, NULL);
|
|
}
|
|
|
|
ao2_cleanup(data);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to send text on PJSIP session */
|
|
static int chan_pjsip_sendtext_data(struct ast_channel *ast, struct ast_msg_data *msg)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct sendtext_data *data = sendtext_data_create(ast, msg);
|
|
|
|
ast_debug(1, "Sending MESSAGE from '%s' to '%s:%s': %s\n",
|
|
ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_FROM),
|
|
ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_TO),
|
|
ast_channel_name(ast),
|
|
ast_msg_data_get_attribute(msg, AST_MSG_DATA_ATTR_BODY));
|
|
|
|
if (!data) {
|
|
return -1;
|
|
}
|
|
|
|
if (ast_sip_push_task(channel->session->serializer, sendtext, data)) {
|
|
ao2_ref(data, -1);
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
|
|
{
|
|
struct ast_msg_data *msg;
|
|
int rc;
|
|
struct ast_msg_data_attribute attrs[] =
|
|
{
|
|
{
|
|
.type = AST_MSG_DATA_ATTR_BODY,
|
|
.value = (char *)text,
|
|
}
|
|
};
|
|
|
|
msg = ast_msg_data_alloc(AST_MSG_DATA_SOURCE_TYPE_UNKNOWN, attrs, ARRAY_LEN(attrs));
|
|
if (!msg) {
|
|
return -1;
|
|
}
|
|
rc = chan_pjsip_sendtext_data(ast, msg);
|
|
ast_free(msg);
|
|
|
|
return rc;
|
|
}
|
|
|
|
static void chan_pjsip_session_begin(struct ast_sip_session *session)
|
|
{
|
|
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (session->endpoint->media.direct_media.glare_mitigation ==
|
|
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
|
|
SCOPE_EXIT_RTN("Direct media no glare mitigation\n");
|
|
}
|
|
|
|
datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
|
|
"direct_media_glare_mitigation");
|
|
|
|
if (!datastore) {
|
|
SCOPE_EXIT_RTN("Couldn't create datastore\n");
|
|
}
|
|
|
|
ast_sip_session_add_datastore(session, datastore);
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
/*! \brief Function called when the session ends */
|
|
static void chan_pjsip_session_end(struct ast_sip_session *session)
|
|
{
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (!session->channel) {
|
|
SCOPE_EXIT_RTN("No channel\n");
|
|
}
|
|
|
|
|
|
if (session->active_media_state &&
|
|
session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
|
|
struct ast_sip_session_media *media =
|
|
session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
|
|
if (media->rtp) {
|
|
ast_rtp_instance_set_stats_vars(session->channel, media->rtp);
|
|
}
|
|
}
|
|
|
|
chan_pjsip_remove_hold(ast_channel_uniqueid(session->channel));
|
|
|
|
ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
|
|
if (!ast_channel_hangupcause(session->channel) && session->inv_session) {
|
|
int cause = ast_sip_hangup_sip2cause(session->inv_session->cause);
|
|
|
|
ast_queue_hangup_with_cause(session->channel, cause);
|
|
} else {
|
|
ast_queue_hangup(session->channel);
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static void set_sipdomain_variable(struct ast_sip_session *session)
|
|
{
|
|
const pj_str_t *host = ast_sip_pjsip_uri_get_hostname(session->request_uri);
|
|
size_t size = pj_strlen(host) + 1;
|
|
char *domain = ast_alloca(size);
|
|
|
|
ast_copy_pj_str(domain, host, size);
|
|
|
|
pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
|
|
return;
|
|
}
|
|
|
|
/*! \brief Function called when a request is received on the session */
|
|
static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
|
|
struct transport_info_data *transport_data;
|
|
pjsip_tx_data *packet = NULL;
|
|
SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (session->channel) {
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s: No channel\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
/* Check for a to-tag to determine if this is a reinvite */
|
|
if (rdata->msg_info.to->tag.slen) {
|
|
/* Weird case. We've received a reinvite but we don't have a channel. The most
|
|
* typical case for this happening is that a blind transfer fails, and so the
|
|
* transferer attempts to reinvite himself back into the call. We already got
|
|
* rid of that channel, and the other side of the call is unrecoverable.
|
|
*
|
|
* We treat this as a failure, so our best bet is to just hang this call
|
|
* up and not create a new channel. Clearing defer_terminate here ensures that
|
|
* calling ast_sip_session_terminate() can result in a BYE being sent ASAP.
|
|
*/
|
|
session->defer_terminate = 0;
|
|
ast_sip_session_terminate(session, 400);
|
|
SCOPE_EXIT_RTN_VALUE(-1, "%s: We have a To tag but no channel. Terminating session\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
|
|
if (!datastore) {
|
|
SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info datastore\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
transport_data = ast_calloc(1, sizeof(*transport_data));
|
|
if (!transport_data) {
|
|
SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc transport_info\n", ast_sip_session_get_name(session));
|
|
}
|
|
pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
|
|
pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
|
|
datastore->data = transport_data;
|
|
ast_sip_session_add_datastore(session, datastore);
|
|
|
|
if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL, NULL))) {
|
|
if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS
|
|
&& packet) {
|
|
ast_sip_session_send_response(session, packet);
|
|
}
|
|
|
|
SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n",
|
|
ast_sip_session_get_name(session));
|
|
}
|
|
|
|
set_sipdomain_variable(session);
|
|
|
|
/* channel gets created on incoming request, but we wait to call start
|
|
so other supplements have a chance to run */
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
static int call_pickup_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
|
|
{
|
|
struct ast_features_pickup_config *pickup_cfg;
|
|
struct ast_channel *chan;
|
|
|
|
/* Check for a to-tag to determine if this is a reinvite */
|
|
if (rdata->msg_info.to->tag.slen) {
|
|
/* We don't care about reinvites */
|
|
return 0;
|
|
}
|
|
|
|
pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
|
|
if (!pickup_cfg) {
|
|
ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n");
|
|
return 0;
|
|
}
|
|
|
|
if (strcmp(session->exten, pickup_cfg->pickupexten)) {
|
|
ao2_ref(pickup_cfg, -1);
|
|
return 0;
|
|
}
|
|
ao2_ref(pickup_cfg, -1);
|
|
|
|
/* We can't directly use session->channel because the pickup operation will cause a masquerade to occur,
|
|
* changing the channel pointer in session to a different channel. To ensure we work on the right channel
|
|
* we store a pointer locally before we begin and keep a reference so it remains valid no matter what.
|
|
*/
|
|
chan = ast_channel_ref(session->channel);
|
|
if (ast_pickup_call(chan)) {
|
|
ast_channel_hangupcause_set(chan, AST_CAUSE_CALL_REJECTED);
|
|
} else {
|
|
ast_channel_hangupcause_set(chan, AST_CAUSE_NORMAL_CLEARING);
|
|
}
|
|
/* A hangup always occurs because the pickup operation will have either failed resulting in the call
|
|
* needing to be hung up OR the pickup operation was a success and the channel we now have is actually
|
|
* the channel that was replaced, which should be hung up since it is literally in limbo not connected
|
|
* to anything at all.
|
|
*/
|
|
ast_hangup(chan);
|
|
ast_channel_unref(chan);
|
|
|
|
return 1;
|
|
}
|
|
|
|
static struct ast_sip_session_supplement call_pickup_supplement = {
|
|
.method = "INVITE",
|
|
.priority = AST_SIP_SUPPLEMENT_PRIORITY_LAST - 1,
|
|
.incoming_request = call_pickup_incoming_request,
|
|
};
|
|
|
|
static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
|
|
{
|
|
int res;
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
/* Check for a to-tag to determine if this is a reinvite */
|
|
if (rdata->msg_info.to->tag.slen) {
|
|
/* We don't care about reinvites */
|
|
SCOPE_EXIT_RTN_VALUE(0, "Reinvite\n");
|
|
}
|
|
|
|
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 PJSIP channel %s\n", ast_channel_name(session->channel));
|
|
|
|
SCOPE_EXIT_RTN_VALUE((res == AST_PBX_SUCCESS) ? 0 : -1, "RC: %d\n", res);
|
|
}
|
|
|
|
static struct ast_sip_session_supplement pbx_start_supplement = {
|
|
.method = "INVITE",
|
|
.priority = AST_SIP_SUPPLEMENT_PRIORITY_LAST,
|
|
.incoming_request = pbx_start_incoming_request,
|
|
};
|
|
|
|
/*! \brief Function called when a response is received on the session */
|
|
static void chan_pjsip_incoming_response_update_cause(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
struct pjsip_status_line status = rdata->msg_info.msg->line.status;
|
|
struct ast_control_pvt_cause_code *cause_code;
|
|
int data_size = sizeof(*cause_code);
|
|
SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
|
|
|
|
if (!session->channel) {
|
|
SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
/* Build and send the tech-specific cause information */
|
|
/* size of the string making up the cause code is "SIP " number + " " + reason length */
|
|
data_size += 4 + 4 + pj_strlen(&status.reason);
|
|
cause_code = ast_alloca(data_size);
|
|
memset(cause_code, 0, data_size);
|
|
|
|
ast_copy_string(cause_code->chan_name, ast_channel_name(session->channel), AST_CHANNEL_NAME);
|
|
|
|
snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %d %.*s", status.code,
|
|
(int) pj_strlen(&status.reason), pj_strbuf(&status.reason));
|
|
|
|
cause_code->ast_cause = ast_sip_hangup_sip2cause(status.code);
|
|
ast_queue_control_data(session->channel, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
|
|
ast_channel_hangupcause_hash_set(session->channel, cause_code, data_size);
|
|
|
|
SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
/*! \brief Function called when a response is received on the session */
|
|
static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
struct pjsip_status_line status = rdata->msg_info.msg->line.status;
|
|
SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
|
|
|
|
if (!session->channel) {
|
|
SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
switch (status.code) {
|
|
case 180: {
|
|
pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
|
|
if (sdp && sdp->body.ptr) {
|
|
ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
|
|
session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
|
|
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
|
|
} else {
|
|
ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
|
|
ast_queue_control(session->channel, AST_CONTROL_RINGING);
|
|
}
|
|
|
|
ast_channel_lock(session->channel);
|
|
if (ast_channel_state(session->channel) != AST_STATE_UP) {
|
|
ast_setstate(session->channel, AST_STATE_RINGING);
|
|
}
|
|
ast_channel_unlock(session->channel);
|
|
break;
|
|
}
|
|
case 183:
|
|
if (session->endpoint->ignore_183_without_sdp) {
|
|
pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
|
|
if (sdp && sdp->body.ptr) {
|
|
ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
|
|
ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS with SDP\n", ast_sip_session_get_name(session),
|
|
(int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
|
|
session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
|
|
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
|
|
}
|
|
} else {
|
|
ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
|
|
ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS without SDP\n", ast_sip_session_get_name(session),
|
|
(int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
|
|
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
|
|
}
|
|
break;
|
|
case 200:
|
|
ast_trace(-1, "%s: Queueing ANSWER\n", ast_sip_session_get_name(session));
|
|
ast_queue_control(session->channel, AST_CONTROL_ANSWER);
|
|
break;
|
|
default:
|
|
ast_trace(-1, "%s: Not queueing anything\n", ast_sip_session_get_name(session));
|
|
break;
|
|
}
|
|
|
|
SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
|
|
if (session->endpoint->media.direct_media.enabled && session->channel) {
|
|
ast_trace(-1, "%s: Queueing SRCCHANGE\n", ast_sip_session_get_name(session));
|
|
ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
|
|
}
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
static int chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (pj_strcmp2(&rdata->msg_info.msg->line.req.method.name, "PRACK") == 0 &&
|
|
pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE) {
|
|
|
|
session->early_confirmed = 1;
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
|
|
}
|
|
|
|
static int update_devstate(void *obj, void *arg, int flags)
|
|
{
|
|
ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE,
|
|
"PJSIP/%s", ast_sorcery_object_get_id(obj));
|
|
return 0;
|
|
}
|
|
|
|
static struct ast_custom_function chan_pjsip_dial_contacts_function = {
|
|
.name = "PJSIP_DIAL_CONTACTS",
|
|
.read = pjsip_acf_dial_contacts_read,
|
|
};
|
|
|
|
static struct ast_custom_function chan_pjsip_parse_uri_function = {
|
|
.name = "PJSIP_PARSE_URI",
|
|
.read = pjsip_acf_parse_uri_read,
|
|
};
|
|
|
|
static struct ast_custom_function media_offer_function = {
|
|
.name = "PJSIP_MEDIA_OFFER",
|
|
.read = pjsip_acf_media_offer_read,
|
|
.write = pjsip_acf_media_offer_write
|
|
};
|
|
|
|
static struct ast_custom_function dtmf_mode_function = {
|
|
.name = "PJSIP_DTMF_MODE",
|
|
.read = pjsip_acf_dtmf_mode_read,
|
|
.write = pjsip_acf_dtmf_mode_write
|
|
};
|
|
|
|
static struct ast_custom_function moh_passthrough_function = {
|
|
.name = "PJSIP_MOH_PASSTHROUGH",
|
|
.read = pjsip_acf_moh_passthrough_read,
|
|
.write = pjsip_acf_moh_passthrough_write
|
|
};
|
|
|
|
static struct ast_custom_function session_refresh_function = {
|
|
.name = "PJSIP_SEND_SESSION_REFRESH",
|
|
.write = pjsip_acf_session_refresh_write,
|
|
};
|
|
|
|
static char *app_pjsip_hangup = "PJSIPHangup";
|
|
|
|
/*!
|
|
* \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)
|
|
{
|
|
struct ao2_container *endpoints;
|
|
|
|
if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
ast_format_cap_append_by_type(chan_pjsip_tech.capabilities, AST_MEDIA_TYPE_AUDIO);
|
|
|
|
ast_rtp_glue_register(&chan_pjsip_rtp_glue);
|
|
|
|
if (ast_channel_register(&chan_pjsip_tech)) {
|
|
ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
|
|
goto end;
|
|
}
|
|
|
|
if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) {
|
|
ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_custom_function_register(&chan_pjsip_parse_uri_function)) {
|
|
ast_log(LOG_ERROR, "Unable to register PJSIP_PARSE_URI dialplan function\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_custom_function_register(&media_offer_function)) {
|
|
ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_custom_function_register(&dtmf_mode_function)) {
|
|
ast_log(LOG_WARNING, "Unable to register PJSIP_DTMF_MODE dialplan function\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_custom_function_register(&moh_passthrough_function)) {
|
|
ast_log(LOG_WARNING, "Unable to register PJSIP_MOH_PASSTHROUGH dialplan function\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_custom_function_register(&session_refresh_function)) {
|
|
ast_log(LOG_WARNING, "Unable to register PJSIP_SEND_SESSION_REFRESH dialplan function\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_register_application_xml(app_pjsip_hangup, pjsip_app_hangup)) {
|
|
ast_log(LOG_WARNING, "Unable to register PJSIPHangup dialplan application\n");
|
|
goto end;
|
|
}
|
|
ast_manager_register_xml(app_pjsip_hangup, EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL, pjsip_action_hangup);
|
|
|
|
|
|
ast_sip_register_service(&refer_callback_module);
|
|
|
|
ast_sip_session_register_supplement(&chan_pjsip_supplement);
|
|
ast_sip_session_register_supplement(&chan_pjsip_supplement_response);
|
|
|
|
if (!(pjsip_uids_onhold = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_RWLOCK,
|
|
AO2_CONTAINER_ALLOC_OPT_DUPS_REJECT, 37, uid_hold_hash_fn,
|
|
uid_hold_sort_fn, NULL))) {
|
|
ast_log(LOG_ERROR, "Unable to create held channels container\n");
|
|
goto end;
|
|
}
|
|
|
|
ast_sip_session_register_supplement(&call_pickup_supplement);
|
|
ast_sip_session_register_supplement(&pbx_start_supplement);
|
|
ast_sip_session_register_supplement(&chan_pjsip_ack_supplement);
|
|
ast_sip_session_register_supplement(&chan_pjsip_prack_supplement);
|
|
|
|
if (pjsip_channel_cli_register()) {
|
|
ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
|
|
goto end;
|
|
}
|
|
|
|
/* since endpoints are loaded before the channel driver their device
|
|
states get set to 'invalid', so they need to be updated */
|
|
if ((endpoints = ast_sip_get_endpoints())) {
|
|
ao2_callback(endpoints, OBJ_NODATA, update_devstate, NULL);
|
|
ao2_ref(endpoints, -1);
|
|
}
|
|
|
|
return 0;
|
|
|
|
end:
|
|
ao2_cleanup(pjsip_uids_onhold);
|
|
pjsip_uids_onhold = NULL;
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_prack_supplement);
|
|
ast_sip_session_unregister_supplement(&pbx_start_supplement);
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
|
|
ast_sip_session_unregister_supplement(&call_pickup_supplement);
|
|
ast_sip_unregister_service(&refer_callback_module);
|
|
ast_custom_function_unregister(&dtmf_mode_function);
|
|
ast_custom_function_unregister(&moh_passthrough_function);
|
|
ast_custom_function_unregister(&media_offer_function);
|
|
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
|
|
ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
|
|
ast_custom_function_unregister(&session_refresh_function);
|
|
ast_unregister_application(app_pjsip_hangup);
|
|
ast_manager_unregister(app_pjsip_hangup);
|
|
|
|
ast_channel_unregister(&chan_pjsip_tech);
|
|
ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
|
|
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
/*! \brief Unload the PJSIP channel from Asterisk */
|
|
static int unload_module(void)
|
|
{
|
|
ao2_cleanup(pjsip_uids_onhold);
|
|
pjsip_uids_onhold = NULL;
|
|
|
|
pjsip_channel_cli_unregister();
|
|
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
|
|
ast_sip_session_unregister_supplement(&pbx_start_supplement);
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_prack_supplement);
|
|
ast_sip_session_unregister_supplement(&call_pickup_supplement);
|
|
|
|
ast_sip_unregister_service(&refer_callback_module);
|
|
|
|
ast_custom_function_unregister(&dtmf_mode_function);
|
|
ast_custom_function_unregister(&moh_passthrough_function);
|
|
ast_custom_function_unregister(&media_offer_function);
|
|
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
|
|
ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
|
|
ast_custom_function_unregister(&session_refresh_function);
|
|
ast_unregister_application(app_pjsip_hangup);
|
|
ast_manager_unregister(app_pjsip_hangup);
|
|
|
|
ast_channel_unregister(&chan_pjsip_tech);
|
|
ao2_ref(chan_pjsip_tech.capabilities, -1);
|
|
ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
|
|
|
|
return 0;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
|
|
.support_level = AST_MODULE_SUPPORT_CORE,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
|
|
.requires = "res_pjsip,res_pjsip_session,res_pjsip_pubsub",
|
|
);
|