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0f7ee5fbcf
This patch introduces a new identifier for channels: tenantid. It's a stringfield on the channel that can be used for general purposes. It will be inherited by other channels the same way that linkedid is. You can set tenantid in a few ways. The first is to set it in the dialplan with the Set and CHANNEL functions: exten => example,1,Set(CHANNEL(tenantid)=My tenant ID) It can also be accessed via CHANNEL: exten => example,2,NoOp(CHANNEL(tenantid)) Another method is to use the new tenantid option for pjsip endpoints in pjsip.conf: [my_endpoint] type=endpoint tenantid=My tenant ID This is considered the best approach since you will be able to see the tenant ID as early as the Newchannel event. It can also be set using set_var in pjsip.conf on the endpoint like setting other channel variable: set_var=CHANNEL(tenantid)=My tenant ID Note that set_var will not show tenant ID on the Newchannel event, however. Tenant ID has also been added to CDR. It's read-only and can be accessed via CDR(tenantid). You can also get the tenant ID of the last channel communicated with via CDR(peertenantid). Tenant ID will also show up in CEL records if it has been set, and the version number has been bumped accordingly. Fixes: #740 UserNote: tenantid has been added to channels. It can be read in dialplan via CHANNEL(tenantid), and it can be set using Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is recommended to use the new tenantid option for pjsip endpoints (e.g., tenantid=My tenant ID) so that it will show up in Newchannel events. You can set it like any other channel variable using set_var in pjsip.conf as well, but note that this will NOT show up in Newchannel events. Tenant ID is also available in CDR and can be accessed with CDR(tenantid). The peer tenant ID can also be accessed with CDR(peertenantid). CEL includes tenant ID as well if it has been set. UpgradeNote: A new versioned struct (ast_channel_initializers) has been added that gets passed to __ast_channel_alloc_ap. The new function ast_channel_alloc_with_initializers should be used when creating channels that require the use of this struct. Currently the only value in the struct is for tenantid, but now more fields can be added to the struct as necessary rather than the __ast_channel_alloc_ap function. A new option (tenantid) has been added to endpoints in pjsip.conf as well. CEL has had its version bumped to include tenant ID.
3435 lines
110 KiB
C
3435 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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|
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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);
|
|
struct ast_sip_session *session;
|
|
int changed = 0;
|
|
int res = 0;
|
|
|
|
/* XXX In an ideal world each media stream would be direct, but for now preserve behavior
|
|
* and connect only the default media sessions for audio and video.
|
|
*/
|
|
|
|
/* The channel needs to be locked when checking for RTP changes.
|
|
* Otherwise, we could end up destroying an underlying RTCP structure
|
|
* at the same time that the channel thread is attempting to read RTCP
|
|
*/
|
|
ast_channel_lock(cdata->chan);
|
|
session = channel->session;
|
|
if (session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO]) {
|
|
changed |= check_for_rtp_changes(
|
|
cdata->chan, cdata->rtp, session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO], session);
|
|
}
|
|
if (session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO]) {
|
|
changed |= check_for_rtp_changes(
|
|
cdata->chan, cdata->vrtp, session->active_media_state->default_session[AST_MEDIA_TYPE_VIDEO], session);
|
|
}
|
|
ast_channel_unlock(cdata->chan);
|
|
|
|
if (direct_media_mitigate_glare(cdata->session)) {
|
|
ast_debug(4, "Disregarding setting RTP on %s: mitigating re-INVITE glare\n", ast_channel_name(cdata->chan));
|
|
ao2_ref(cdata, -1);
|
|
return 0;
|
|
}
|
|
|
|
if (cdata->cap && ast_format_cap_count(cdata->cap) &&
|
|
!ast_format_cap_identical(cdata->session->direct_media_cap, cdata->cap)) {
|
|
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;
|
|
struct ast_channel_initializers initializers = {
|
|
.version = AST_CHANNEL_INITIALIZERS_VERSION,
|
|
.tenantid = session->endpoint->tenantid,
|
|
};
|
|
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_initializers(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, &initializers, "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 = S_OR(ast_channel_macrocontext(ast), 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",
|
|
);
|