Testsuite tests will occasionally fail because on reception of a 200 OK SIP response, an AST_CONTROL_ANSWER frame is queued prior to when media has finished being negotiated. This is because session supplements are called into before PJSIP's inv_session code has told us that media has been updated. Sometimes the queued answer frame is handled by the PBX thread before the ensuing media negotiations occur, causing a test failure. As it turns out, there is another place that session supplements could be called into, which is after media has finished getting negotiated. What this commit introduces is a means for session supplements to indicate when they wish to be called into when handling an incoming SIP response. By default, all session supplements will be run at the same point that they were prior to this commit. However, session supplements may indicate that they wish to be handled earlier than normal on redirects, or they may indicate they wish to be handled after media has been negotiated. In this changeset, two session supplements have been updated to indicate a preference for when they should be run: res_pjsip_diversion executes before handling redirection in order to get information from the Diversion header, and chan_pjsip now handles responses to INVITEs after media negotiation to fix the race condition mentioned previously. ASTERISK-24212 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3930 ........ Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422542 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2313 lines
69 KiB
C
2313 lines
69 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_session</depend>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <pjsip.h>
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#include <pjsip_ua.h>
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#include <pjlib.h>
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/lock.h"
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#include "asterisk/channel.h"
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#include "asterisk/module.h"
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#include "asterisk/pbx.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/acl.h"
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#include "asterisk/callerid.h"
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#include "asterisk/file.h"
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#include "asterisk/cli.h"
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#include "asterisk/app.h"
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#include "asterisk/musiconhold.h"
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#include "asterisk/causes.h"
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#include "asterisk/taskprocessor.h"
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#include "asterisk/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/res_pjsip.h"
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#include "asterisk/res_pjsip_session.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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AST_THREADSTORAGE(uniqueid_threadbuf);
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#define UNIQUEID_BUFSIZE 256
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static const char desc[] = "PJSIP Channel";
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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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struct chan_pjsip_pvt *pvt = obj;
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int i;
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for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
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ao2_cleanup(pvt->media[i]);
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pvt->media[i] = NULL;
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}
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}
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/* \brief Asterisk core interaction functions */
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static struct ast_channel *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 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(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_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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.send_text = chan_pjsip_sendtext,
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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 = chan_pjsip_read,
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.write = chan_pjsip_write,
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.write_video = chan_pjsip_write,
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.exception = chan_pjsip_read,
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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
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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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/*! \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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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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/*! \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 chan_pjsip_pvt *pvt = channel->pvt;
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struct ast_sip_endpoint *endpoint;
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if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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endpoint = channel->session->endpoint;
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*instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
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ao2_ref(*instance, +1);
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ast_assert(endpoint != NULL);
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if (endpoint->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 chan_pjsip_pvt *pvt = channel->pvt;
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struct ast_sip_endpoint *endpoint;
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if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
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return AST_RTP_GLUE_RESULT_FORBID;
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}
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endpoint = channel->session->endpoint;
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*instance = pvt->media[SIP_MEDIA_VIDEO]->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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struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
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ast_format_cap_append_from_cap(result, channel->session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
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}
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static int send_direct_media_request(void *data)
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{
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RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
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return ast_sip_session_refresh(session, NULL, NULL, NULL,
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session->endpoint->media.direct_media.method, 1);
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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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static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp,
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struct ast_sip_session_media *media, int rtcp_fd)
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{
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int changed = 0;
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if (rtp) {
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changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
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if (media->rtp) {
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ast_channel_set_fd(chan, rtcp_fd, -1);
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ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
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}
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} else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
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ast_sockaddr_setnull(&media->direct_media_addr);
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changed = 1;
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if (media->rtp) {
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ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
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ast_channel_set_fd(chan, rtcp_fd, ast_rtp_instance_fd(media->rtp, 1));
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}
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}
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return changed;
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}
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/*! \brief Function called by RTP engine to change where the remote party should send media */
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static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
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struct ast_rtp_instance *rtp,
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struct ast_rtp_instance *vrtp,
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struct ast_rtp_instance *tpeer,
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const struct ast_format_cap *cap,
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int nat_active)
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{
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struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
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struct chan_pjsip_pvt *pvt = channel->pvt;
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struct ast_sip_session *session = channel->session;
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int changed = 0;
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/* Don't try to do any direct media shenanigans on early bridges */
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if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) {
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ast_debug(4, "Disregarding setting RTP on %s: channel is not bridged\n", ast_channel_name(chan));
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return 0;
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}
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if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
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ast_debug(4, "Disregarding setting RTP on %s: NAT is active\n", ast_channel_name(chan));
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return 0;
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}
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if (pvt->media[SIP_MEDIA_AUDIO]) {
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changed |= check_for_rtp_changes(chan, rtp, pvt->media[SIP_MEDIA_AUDIO], 1);
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}
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if (pvt->media[SIP_MEDIA_VIDEO]) {
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changed |= check_for_rtp_changes(chan, vrtp, pvt->media[SIP_MEDIA_VIDEO], 3);
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}
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if (direct_media_mitigate_glare(session)) {
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ast_debug(4, "Disregarding setting RTP on %s: mitigating re-INVITE glare\n", ast_channel_name(chan));
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return 0;
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}
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if (cap && ast_format_cap_count(cap) && !ast_format_cap_identical(session->direct_media_cap, cap)) {
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ast_format_cap_remove_by_type(session->direct_media_cap, AST_MEDIA_TYPE_UNKNOWN);
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ast_format_cap_append_from_cap(session->direct_media_cap, cap, AST_MEDIA_TYPE_UNKNOWN);
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changed = 1;
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}
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if (changed) {
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ao2_ref(session, +1);
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ast_debug(4, "RTP changed on %s; initiating direct media update\n", ast_channel_name(chan));
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if (ast_sip_push_task(session->serializer, send_direct_media_request, session)) {
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ao2_cleanup(session);
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}
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}
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return 0;
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}
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/*! \brief Local glue for interacting with the RTP engine core */
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static struct ast_rtp_glue chan_pjsip_rtp_glue = {
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.type = "PJSIP",
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.get_rtp_info = chan_pjsip_get_rtp_peer,
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.get_vrtp_info = chan_pjsip_get_vrtp_peer,
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.get_codec = chan_pjsip_get_codec,
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.update_peer = chan_pjsip_set_rtp_peer,
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};
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static void set_channel_on_rtp_instance(struct chan_pjsip_pvt *pvt, const char *channel_id)
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{
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if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
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ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, channel_id);
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}
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if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
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ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, channel_id);
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}
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}
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|
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/*! \brief Function called to create a new PJSIP Asterisk channel */
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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)
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{
|
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struct ast_channel *chan;
|
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struct ast_format_cap *caps;
|
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struct ast_format *fmt;
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RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
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struct ast_sip_channel_pvt *channel;
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struct ast_variable *var;
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|
|
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if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
|
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return NULL;
|
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}
|
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caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
|
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if (!caps) {
|
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return NULL;
|
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}
|
|
|
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chan = ast_channel_alloc_with_endpoint(1, state,
|
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S_COR(session->id.number.valid, session->id.number.str, ""),
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S_COR(session->id.name.valid, session->id.name.str, ""),
|
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session->endpoint->accountcode, "", "", assignedids, requestor, 0,
|
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session->endpoint->persistent, "PJSIP/%s-%08x",
|
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ast_sorcery_object_get_id(session->endpoint),
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(unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
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if (!chan) {
|
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ao2_ref(caps, -1);
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return NULL;
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}
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|
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ast_channel_tech_set(chan, &chan_pjsip_tech);
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|
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if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
|
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ao2_ref(caps, -1);
|
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ast_channel_unlock(chan);
|
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ast_hangup(chan);
|
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return NULL;
|
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}
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|
|
|
ast_channel_stage_snapshot(chan);
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|
|
|
ast_channel_tech_pvt_set(chan, channel);
|
|
|
|
if (!ast_format_cap_count(session->req_caps) ||
|
|
!ast_format_cap_iscompatible(session->req_caps, session->endpoint->media.codecs)) {
|
|
ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
|
|
} else {
|
|
ast_format_cap_append_from_cap(caps, session->req_caps, AST_MEDIA_TYPE_UNKNOWN);
|
|
}
|
|
|
|
ast_channel_nativeformats_set(chan, caps);
|
|
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_context_set(chan, session->endpoint->context);
|
|
ast_channel_exten_set(chan, S_OR(exten, "s"));
|
|
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);
|
|
|
|
/* If res_pjsip_session is ever updated to create/destroy ast_sip_session_media
|
|
* during a call such as if multiple same-type stream support is introduced,
|
|
* these will need to be recaptured as well */
|
|
pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
|
|
pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
|
|
set_channel_on_rtp_instance(pvt, ast_channel_uniqueid(chan));
|
|
|
|
return chan;
|
|
}
|
|
|
|
static int answer(void *data)
|
|
{
|
|
pj_status_t status = PJ_SUCCESS;
|
|
pjsip_tx_data *packet = NULL;
|
|
struct ast_sip_session *session = data;
|
|
|
|
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);
|
|
}
|
|
|
|
ao2_ref(session, -1);
|
|
|
|
return (status == PJ_SUCCESS) ? 0 : -1;
|
|
}
|
|
|
|
/*! \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);
|
|
|
|
if (ast_channel_state(ast) == AST_STATE_UP) {
|
|
return 0;
|
|
}
|
|
|
|
ast_setstate(ast, AST_STATE_UP);
|
|
|
|
ao2_ref(channel->session, +1);
|
|
if (ast_sip_push_task(channel->session->serializer, answer, channel->session)) {
|
|
ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
|
|
ao2_cleanup(channel->session);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Internal helper function called when CNG tone is detected */
|
|
static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
|
|
{
|
|
const char *target_context;
|
|
int exists;
|
|
|
|
/* If we only needed this DSP for fax detection purposes we can just drop it now */
|
|
if (session->endpoint->dtmf == AST_SIP_DTMF_INBAND) {
|
|
ast_dsp_set_features(session->dsp, DSP_FEATURE_DIGIT_DETECT);
|
|
} 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(session->channel), "fax")) {
|
|
return f;
|
|
}
|
|
|
|
target_context = S_OR(ast_channel_macrocontext(session->channel), ast_channel_context(session->channel));
|
|
|
|
/* 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_channel_unlock(session->channel);
|
|
exists = ast_exists_extension(session->channel, target_context, "fax", 1,
|
|
S_COR(ast_channel_caller(session->channel)->id.number.valid,
|
|
ast_channel_caller(session->channel)->id.number.str, NULL));
|
|
ast_channel_lock(session->channel);
|
|
|
|
if (exists) {
|
|
ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n",
|
|
ast_channel_name(session->channel));
|
|
pbx_builtin_setvar_helper(session->channel, "FAXEXTEN", ast_channel_exten(session->channel));
|
|
if (ast_async_goto(session->channel, target_context, "fax", 1)) {
|
|
ast_log(LOG_ERROR, "Failed to async goto '%s' into fax extension in '%s'\n",
|
|
ast_channel_name(session->channel), target_context);
|
|
}
|
|
ast_frfree(f);
|
|
f = &ast_null_frame;
|
|
} else {
|
|
ast_log(LOG_NOTICE, "FAX CNG detected on '%s' but no fax extension in '%s'\n",
|
|
ast_channel_name(session->channel), target_context);
|
|
}
|
|
|
|
return f;
|
|
}
|
|
|
|
/*! \brief Function called by core to read any waiting frames */
|
|
static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct chan_pjsip_pvt *pvt = channel->pvt;
|
|
struct ast_frame *f;
|
|
struct ast_sip_session_media *media = NULL;
|
|
int rtcp = 0;
|
|
int fdno = ast_channel_fdno(ast);
|
|
|
|
switch (fdno) {
|
|
case 0:
|
|
media = pvt->media[SIP_MEDIA_AUDIO];
|
|
break;
|
|
case 1:
|
|
media = pvt->media[SIP_MEDIA_AUDIO];
|
|
rtcp = 1;
|
|
break;
|
|
case 2:
|
|
media = pvt->media[SIP_MEDIA_VIDEO];
|
|
break;
|
|
case 3:
|
|
media = pvt->media[SIP_MEDIA_VIDEO];
|
|
rtcp = 1;
|
|
break;
|
|
}
|
|
|
|
if (!media || !media->rtp) {
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
if (!(f = ast_rtp_instance_read(media->rtp, rtcp))) {
|
|
return f;
|
|
}
|
|
|
|
if (f->frametype != AST_FRAME_VOICE) {
|
|
return f;
|
|
}
|
|
|
|
if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
|
|
struct ast_format_cap *caps;
|
|
|
|
ast_debug(1, "Oooh, format changed to %s\n", ast_format_get_name(f->subclass.format));
|
|
|
|
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
|
|
if (caps) {
|
|
ast_format_cap_append(caps, f->subclass.format, 0);
|
|
ast_channel_nativeformats_set(ast, caps);
|
|
ao2_ref(caps, -1);
|
|
}
|
|
|
|
ast_set_read_format(ast, ast_channel_readformat(ast));
|
|
ast_set_write_format(ast, ast_channel_writeformat(ast));
|
|
}
|
|
|
|
if (channel->session->dsp) {
|
|
f = ast_dsp_process(ast, channel->session->dsp, f);
|
|
|
|
if (f && (f->frametype == AST_FRAME_DTMF)) {
|
|
if (f->subclass.integer == 'f') {
|
|
ast_debug(3, "Fax CNG detected on %s\n", ast_channel_name(ast));
|
|
f = chan_pjsip_cng_tone_detected(channel->session, f);
|
|
} else {
|
|
ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
|
|
ast_channel_name(ast));
|
|
}
|
|
}
|
|
}
|
|
|
|
return f;
|
|
}
|
|
|
|
/*! \brief Function called by core to write frames */
|
|
static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct chan_pjsip_pvt *pvt = channel->pvt;
|
|
struct ast_sip_session_media *media;
|
|
int res = 0;
|
|
|
|
switch (frame->frametype) {
|
|
case AST_FRAME_VOICE:
|
|
media = pvt->media[SIP_MEDIA_AUDIO];
|
|
|
|
if (!media) {
|
|
return 0;
|
|
}
|
|
if (ast_format_cap_iscompatible_format(ast_channel_nativeformats(ast), frame->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
|
|
struct ast_str *cap_buf = ast_str_alloca(128);
|
|
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;
|
|
}
|
|
if (media->rtp) {
|
|
res = ast_rtp_instance_write(media->rtp, frame);
|
|
}
|
|
break;
|
|
case AST_FRAME_VIDEO:
|
|
if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
|
|
res = ast_rtp_instance_write(media->rtp, frame);
|
|
}
|
|
break;
|
|
case AST_FRAME_MODEM:
|
|
break;
|
|
default:
|
|
ast_log(LOG_WARNING, "Can't send %u type frames with PJSIP\n", frame->frametype);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
struct fixup_data {
|
|
struct ast_sip_session *session;
|
|
struct ast_channel *chan;
|
|
};
|
|
|
|
static int fixup(void *data)
|
|
{
|
|
struct fixup_data *fix_data = data;
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
|
|
struct chan_pjsip_pvt *pvt = channel->pvt;
|
|
|
|
channel->session->channel = fix_data->chan;
|
|
set_channel_on_rtp_instance(pvt, ast_channel_uniqueid(fix_data->chan));
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \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);
|
|
struct fixup_data fix_data;
|
|
|
|
fix_data.session = channel->session;
|
|
fix_data.chan = newchan;
|
|
|
|
if (channel->session->channel != oldchan) {
|
|
return -1;
|
|
}
|
|
|
|
if (ast_sip_push_task_synchronous(channel->session->serializer, fixup, &fix_data)) {
|
|
ast_log(LOG_WARNING, "Unable to perform channel fixup\n");
|
|
return -1;
|
|
}
|
|
|
|
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);
|
|
RAII_VAR(struct stasis_cache *, cache, 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 || !(cache = ast_channel_cache())) {
|
|
return state;
|
|
}
|
|
|
|
ast_devstate_aggregate_init(&aggregate);
|
|
|
|
ao2_ref(cache, +1);
|
|
|
|
for (num = 0; num < endpoint_snapshot->num_channels; num++) {
|
|
RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
|
|
struct ast_channel_snapshot *snapshot;
|
|
|
|
msg = stasis_cache_get(cache, ast_channel_snapshot_type(),
|
|
endpoint_snapshot->channel_ids[num]);
|
|
|
|
if (!msg) {
|
|
continue;
|
|
}
|
|
|
|
snapshot = stasis_message_data(msg);
|
|
|
|
if (chan_pjsip_get_hold(snapshot->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++;
|
|
}
|
|
}
|
|
|
|
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);
|
|
struct ast_sip_session *session = channel->session;
|
|
int res = -1;
|
|
enum ast_sip_session_t38state state = T38_STATE_UNAVAILABLE;
|
|
|
|
switch (option) {
|
|
case AST_OPTION_T38_STATE:
|
|
if (session->endpoint->media.t38.enabled) {
|
|
switch (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 (!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 (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 (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;
|
|
}
|
|
|
|
/*! \brief Update connected line information */
|
|
static int update_connected_line_information(void *data)
|
|
{
|
|
RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
|
|
|
|
if ((ast_channel_state(session->channel) != AST_STATE_UP) && (session->inv_session->role == PJSIP_UAS_ROLE)) {
|
|
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);
|
|
}
|
|
}
|
|
} else {
|
|
enum ast_sip_session_refresh_method method = session->endpoint->id.refresh_method;
|
|
struct ast_party_id connected_id;
|
|
|
|
if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
|
|
method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
|
|
}
|
|
|
|
/*
|
|
* We can get away with a shallow copy here because we are
|
|
* not looking at strings.
|
|
*/
|
|
ast_channel_lock(session->channel);
|
|
connected_id = ast_channel_connected_effective_id(session->channel);
|
|
ast_channel_unlock(session->channel);
|
|
|
|
if ((session->endpoint->id.send_pai || session->endpoint->id.send_rpid) &&
|
|
(session->endpoint->id.trust_outbound ||
|
|
((connected_id.name.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED &&
|
|
(connected_id.number.presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED))) {
|
|
ast_sip_session_refresh(session, NULL, NULL, NULL, method, 1);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \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 chan_pjsip_pvt *pvt = channel->pvt;
|
|
struct ast_sip_session_media *media;
|
|
int response_code = 0;
|
|
int res = 0;
|
|
char *device_buf;
|
|
size_t device_buf_size;
|
|
|
|
switch (condition) {
|
|
case AST_CONTROL_RINGING:
|
|
if (ast_channel_state(ast) == AST_STATE_RING) {
|
|
if (channel->session->endpoint->inband_progress) {
|
|
response_code = 183;
|
|
res = -1;
|
|
} 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;
|
|
}
|
|
break;
|
|
case AST_CONTROL_VIDUPDATE:
|
|
media = pvt->media[SIP_MEDIA_VIDEO];
|
|
if (media && 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) {
|
|
/* 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;
|
|
}
|
|
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_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);
|
|
ast_moh_start(ast, data, NULL);
|
|
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);
|
|
ast_moh_stop(ast);
|
|
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 -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 || ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
|
|
ast_log(LOG_NOTICE, "Cannot send response code %d to endpoint %s. Could not queue task properly\n",
|
|
response_code, ast_sorcery_object_get_id(channel->session->endpoint));
|
|
ao2_cleanup(ind_data);
|
|
res = -1;
|
|
}
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
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) {
|
|
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))) {
|
|
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));
|
|
}
|
|
|
|
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;
|
|
|
|
if (pjsip_xfer_create_uac(session->inv_session->dlg, NULL, &sub) != PJ_SUCCESS) {
|
|
message = AST_TRANSFER_FAILED;
|
|
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
|
|
return;
|
|
}
|
|
|
|
if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
|
|
message = AST_TRANSFER_FAILED;
|
|
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
pjsip_evsub_terminate(sub, PJ_FALSE);
|
|
|
|
return;
|
|
}
|
|
|
|
pjsip_xfer_send_request(sub, packet);
|
|
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
|
|
}
|
|
|
|
static int transfer(void *data)
|
|
{
|
|
struct transfer_data *trnf_data = data;
|
|
|
|
if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
|
|
transfer_redirect(trnf_data->session, trnf_data->target);
|
|
} else {
|
|
transfer_refer(trnf_data->session, trnf_data->target);
|
|
}
|
|
|
|
ao2_ref(trnf_data, -1);
|
|
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 chan_pjsip_pvt *pvt = channel->pvt;
|
|
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
|
|
int res = 0;
|
|
|
|
switch (channel->session->endpoint->dtmf) {
|
|
case AST_SIP_DTMF_RFC_4733:
|
|
if (!media || !media->rtp) {
|
|
return -1;
|
|
}
|
|
|
|
ast_rtp_instance_dtmf_begin(media->rtp, digit);
|
|
case AST_SIP_DTMF_NONE:
|
|
break;
|
|
case AST_SIP_DTMF_INBAND:
|
|
res = -1;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
struct info_dtmf_data {
|
|
struct ast_sip_session *session;
|
|
char digit;
|
|
unsigned int duration;
|
|
};
|
|
|
|
static void info_dtmf_data_destroy(void *obj)
|
|
{
|
|
struct info_dtmf_data *dtmf_data = obj;
|
|
ao2_ref(dtmf_data->session, -1);
|
|
}
|
|
|
|
static struct info_dtmf_data *info_dtmf_data_alloc(struct ast_sip_session *session, char digit, unsigned int duration)
|
|
{
|
|
struct info_dtmf_data *dtmf_data = ao2_alloc(sizeof(*dtmf_data), info_dtmf_data_destroy);
|
|
if (!dtmf_data) {
|
|
return NULL;
|
|
}
|
|
ao2_ref(session, +1);
|
|
dtmf_data->session = session;
|
|
dtmf_data->digit = digit;
|
|
dtmf_data->duration = duration;
|
|
return dtmf_data;
|
|
}
|
|
|
|
static int transmit_info_dtmf(void *data)
|
|
{
|
|
RAII_VAR(struct info_dtmf_data *, dtmf_data, data, ao2_cleanup);
|
|
|
|
struct ast_sip_session *session = dtmf_data->session;
|
|
struct pjsip_tx_data *tdata;
|
|
|
|
RAII_VAR(struct ast_str *, body_text, NULL, ast_free_ptr);
|
|
|
|
struct ast_sip_body body = {
|
|
.type = "application",
|
|
.subtype = "dtmf-relay",
|
|
};
|
|
|
|
if (!(body_text = ast_str_create(32))) {
|
|
ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
|
|
return -1;
|
|
}
|
|
ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
|
|
|
|
body.body_text = ast_str_buffer(body_text);
|
|
|
|
if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, 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 chan_pjsip_pvt *pvt = channel->pvt;
|
|
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
|
|
int res = 0;
|
|
|
|
switch (channel->session->endpoint->dtmf) {
|
|
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 -1;
|
|
}
|
|
|
|
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
|
|
case AST_SIP_DTMF_NONE:
|
|
break;
|
|
case AST_SIP_DTMF_INBAND:
|
|
res = -1;
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static 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;
|
|
struct chan_pjsip_pvt *pvt = channel->pvt;
|
|
pjsip_tx_data *tdata;
|
|
|
|
int 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(pvt, ast_channel_uniqueid(session->channel));
|
|
update_initial_connected_line(session);
|
|
ast_sip_session_send_request(session, tdata);
|
|
}
|
|
ao2_ref(channel, -1);
|
|
return 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);
|
|
|
|
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);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
|
|
static int hangup_cause2sip(int cause)
|
|
{
|
|
switch (cause) {
|
|
case AST_CAUSE_UNALLOCATED: /* 1 */
|
|
case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
|
|
case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
|
|
return 404;
|
|
case AST_CAUSE_CONGESTION: /* 34 */
|
|
case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
|
|
return 503;
|
|
case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
|
|
return 408;
|
|
case AST_CAUSE_NO_ANSWER: /* 19 */
|
|
case AST_CAUSE_UNREGISTERED: /* 20 */
|
|
return 480;
|
|
case AST_CAUSE_CALL_REJECTED: /* 21 */
|
|
return 403;
|
|
case AST_CAUSE_NUMBER_CHANGED: /* 22 */
|
|
return 410;
|
|
case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
|
|
return 480;
|
|
case AST_CAUSE_INVALID_NUMBER_FORMAT:
|
|
return 484;
|
|
case AST_CAUSE_USER_BUSY:
|
|
return 486;
|
|
case AST_CAUSE_FAILURE:
|
|
return 500;
|
|
case AST_CAUSE_FACILITY_REJECTED: /* 29 */
|
|
return 501;
|
|
case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
|
|
return 503;
|
|
case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
|
|
return 502;
|
|
case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
|
|
return 488;
|
|
case AST_CAUSE_INTERWORKING: /* Unspecified Interworking issues */
|
|
return 500;
|
|
case AST_CAUSE_NOTDEFINED:
|
|
default:
|
|
ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
|
|
return 0;
|
|
}
|
|
|
|
/* Never reached */
|
|
return 0;
|
|
}
|
|
|
|
struct hangup_data {
|
|
int cause;
|
|
struct ast_channel *chan;
|
|
};
|
|
|
|
static void hangup_data_destroy(void *obj)
|
|
{
|
|
struct hangup_data *h_data = obj;
|
|
|
|
h_data->chan = ast_channel_unref(h_data->chan);
|
|
}
|
|
|
|
static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
|
|
{
|
|
struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
|
|
|
|
if (!h_data) {
|
|
return NULL;
|
|
}
|
|
|
|
h_data->cause = cause;
|
|
h_data->chan = ast_channel_ref(chan);
|
|
|
|
return h_data;
|
|
}
|
|
|
|
/*! \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, struct chan_pjsip_pvt *pvt)
|
|
{
|
|
session->channel = NULL;
|
|
set_channel_on_rtp_instance(pvt, "");
|
|
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);
|
|
struct chan_pjsip_pvt *pvt = channel->pvt;
|
|
struct ast_sip_session *session = channel->session;
|
|
int cause = h_data->cause;
|
|
|
|
if (!session->defer_terminate) {
|
|
pj_status_t status;
|
|
pjsip_tx_data *packet = NULL;
|
|
|
|
if (session->inv_session->state == PJSIP_INV_STATE_NULL) {
|
|
pjsip_inv_terminate(session->inv_session, cause ? cause : 603, PJ_TRUE);
|
|
} else if (((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS)
|
|
&& packet) {
|
|
if (packet->msg->type == PJSIP_RESPONSE_MSG) {
|
|
ast_sip_session_send_response(session, packet);
|
|
} else {
|
|
ast_sip_session_send_request(session, packet);
|
|
}
|
|
}
|
|
}
|
|
|
|
clear_session_and_channel(session, ast, pvt);
|
|
ao2_cleanup(channel);
|
|
ao2_cleanup(h_data);
|
|
|
|
return 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);
|
|
struct chan_pjsip_pvt *pvt = channel->pvt;
|
|
int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
|
|
struct hangup_data *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;
|
|
}
|
|
|
|
return 0;
|
|
|
|
failure:
|
|
/* Go ahead and do our cleanup of the session and channel even if we're not going
|
|
* to be able to send our SIP request/response
|
|
*/
|
|
clear_session_and_channel(channel->session, ast, pvt);
|
|
ao2_cleanup(channel);
|
|
ao2_cleanup(h_data);
|
|
|
|
return -1;
|
|
}
|
|
|
|
struct request_data {
|
|
struct ast_sip_session *session;
|
|
struct ast_format_cap *caps;
|
|
const char *dest;
|
|
int cause;
|
|
};
|
|
|
|
static int request(void *obj)
|
|
{
|
|
struct request_data *req_data = obj;
|
|
struct ast_sip_session *session = NULL;
|
|
char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
|
|
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
|
|
|
|
AST_DECLARE_APP_ARGS(args,
|
|
AST_APP_ARG(endpoint);
|
|
AST_APP_ARG(aor);
|
|
);
|
|
|
|
if (ast_strlen_zero(tmp)) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
|
|
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
|
|
return -1;
|
|
}
|
|
|
|
AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
|
|
|
|
/* If a request user has been specified extract it from the endpoint name portion */
|
|
if ((endpoint_name = strchr(args.endpoint, '@'))) {
|
|
request_user = args.endpoint;
|
|
*endpoint_name++ = '\0';
|
|
} else {
|
|
endpoint_name = args.endpoint;
|
|
}
|
|
|
|
if (ast_strlen_zero(endpoint_name)) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
|
|
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
|
|
} else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
|
|
ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
|
|
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
return -1;
|
|
}
|
|
|
|
if (!(session = ast_sip_session_create_outgoing(endpoint, NULL, args.aor, request_user, req_data->caps))) {
|
|
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
return -1;
|
|
}
|
|
|
|
req_data->session = session;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to create a new outgoing PJSIP session */
|
|
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 request_data req_data;
|
|
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
|
|
|
|
req_data.caps = cap;
|
|
req_data.dest = data;
|
|
|
|
if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
|
|
*cause = req_data.cause;
|
|
return NULL;
|
|
}
|
|
|
|
session = req_data.session;
|
|
|
|
if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, assignedids, requestor, NULL))) {
|
|
/* Session needs to be terminated prematurely */
|
|
return NULL;
|
|
}
|
|
|
|
return session->channel;
|
|
}
|
|
|
|
struct sendtext_data {
|
|
struct ast_sip_session *session;
|
|
char text[0];
|
|
};
|
|
|
|
static void sendtext_data_destroy(void *obj)
|
|
{
|
|
struct sendtext_data *data = obj;
|
|
ao2_ref(data->session, -1);
|
|
}
|
|
|
|
static struct sendtext_data* sendtext_data_create(struct ast_sip_session *session, const char *text)
|
|
{
|
|
int size = strlen(text) + 1;
|
|
struct sendtext_data *data = ao2_alloc(sizeof(*data)+size, sendtext_data_destroy);
|
|
|
|
if (!data) {
|
|
return NULL;
|
|
}
|
|
|
|
data->session = session;
|
|
ao2_ref(data->session, +1);
|
|
ast_copy_string(data->text, text, size);
|
|
return data;
|
|
}
|
|
|
|
static int sendtext(void *obj)
|
|
{
|
|
RAII_VAR(struct sendtext_data *, data, obj, ao2_cleanup);
|
|
pjsip_tx_data *tdata;
|
|
|
|
const struct ast_sip_body body = {
|
|
.type = "text",
|
|
.subtype = "plain",
|
|
.body_text = data->text
|
|
};
|
|
|
|
/* NOT ast_strlen_zero, because a zero-length message is specifically
|
|
* allowed by RFC 3428 (See section 10, Examples) */
|
|
if (!data->text) {
|
|
return 0;
|
|
}
|
|
|
|
ast_debug(3, "Sending in dialog SIP message\n");
|
|
|
|
ast_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, NULL, &tdata);
|
|
ast_sip_add_body(tdata, &body);
|
|
ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint, NULL, NULL);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called by core to send text on PJSIP session */
|
|
static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
|
|
struct sendtext_data *data = sendtext_data_create(channel->session, text);
|
|
|
|
if (!data || ast_sip_push_task(channel->session->serializer, sendtext, data)) {
|
|
ao2_ref(data, -1);
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
|
|
static int hangup_sip2cause(int cause)
|
|
{
|
|
/* Possible values taken from causes.h */
|
|
|
|
switch(cause) {
|
|
case 401: /* Unauthorized */
|
|
return AST_CAUSE_CALL_REJECTED;
|
|
case 403: /* Not found */
|
|
return AST_CAUSE_CALL_REJECTED;
|
|
case 404: /* Not found */
|
|
return AST_CAUSE_UNALLOCATED;
|
|
case 405: /* Method not allowed */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 407: /* Proxy authentication required */
|
|
return AST_CAUSE_CALL_REJECTED;
|
|
case 408: /* No reaction */
|
|
return AST_CAUSE_NO_USER_RESPONSE;
|
|
case 409: /* Conflict */
|
|
return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
|
|
case 410: /* Gone */
|
|
return AST_CAUSE_NUMBER_CHANGED;
|
|
case 411: /* Length required */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 413: /* Request entity too large */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 414: /* Request URI too large */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 415: /* Unsupported media type */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 420: /* Bad extension */
|
|
return AST_CAUSE_NO_ROUTE_DESTINATION;
|
|
case 480: /* No answer */
|
|
return AST_CAUSE_NO_ANSWER;
|
|
case 481: /* No answer */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 482: /* Loop detected */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 483: /* Too many hops */
|
|
return AST_CAUSE_NO_ANSWER;
|
|
case 484: /* Address incomplete */
|
|
return AST_CAUSE_INVALID_NUMBER_FORMAT;
|
|
case 485: /* Ambiguous */
|
|
return AST_CAUSE_UNALLOCATED;
|
|
case 486: /* Busy everywhere */
|
|
return AST_CAUSE_BUSY;
|
|
case 487: /* Request terminated */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 488: /* No codecs approved */
|
|
return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
|
|
case 491: /* Request pending */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 493: /* Undecipherable */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 500: /* Server internal failure */
|
|
return AST_CAUSE_FAILURE;
|
|
case 501: /* Call rejected */
|
|
return AST_CAUSE_FACILITY_REJECTED;
|
|
case 502:
|
|
return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
|
|
case 503: /* Service unavailable */
|
|
return AST_CAUSE_CONGESTION;
|
|
case 504: /* Gateway timeout */
|
|
return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
|
|
case 505: /* SIP version not supported */
|
|
return AST_CAUSE_INTERWORKING;
|
|
case 600: /* Busy everywhere */
|
|
return AST_CAUSE_USER_BUSY;
|
|
case 603: /* Decline */
|
|
return AST_CAUSE_CALL_REJECTED;
|
|
case 604: /* Does not exist anywhere */
|
|
return AST_CAUSE_UNALLOCATED;
|
|
case 606: /* Not acceptable */
|
|
return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
|
|
default:
|
|
if (cause < 500 && cause >= 400) {
|
|
/* 4xx class error that is unknown - someting wrong with our request */
|
|
return AST_CAUSE_INTERWORKING;
|
|
} else if (cause < 600 && cause >= 500) {
|
|
/* 5xx class error - problem in the remote end */
|
|
return AST_CAUSE_CONGESTION;
|
|
} else if (cause < 700 && cause >= 600) {
|
|
/* 6xx - global errors in the 4xx class */
|
|
return AST_CAUSE_INTERWORKING;
|
|
}
|
|
return AST_CAUSE_NORMAL;
|
|
}
|
|
/* Never reached */
|
|
return 0;
|
|
}
|
|
|
|
static void chan_pjsip_session_begin(struct ast_sip_session *session)
|
|
{
|
|
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
|
|
|
|
if (session->endpoint->media.direct_media.glare_mitigation ==
|
|
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_NONE) {
|
|
return;
|
|
}
|
|
|
|
datastore = ast_sip_session_alloc_datastore(&direct_media_mitigation_info,
|
|
"direct_media_glare_mitigation");
|
|
|
|
if (!datastore) {
|
|
return;
|
|
}
|
|
|
|
ast_sip_session_add_datastore(session, datastore);
|
|
}
|
|
|
|
/*! \brief Function called when the session ends */
|
|
static void chan_pjsip_session_end(struct ast_sip_session *session)
|
|
{
|
|
if (!session->channel) {
|
|
return;
|
|
}
|
|
|
|
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 = hangup_sip2cause(session->inv_session->cause);
|
|
|
|
ast_queue_hangup_with_cause(session->channel, cause);
|
|
} else {
|
|
ast_queue_hangup(session->channel);
|
|
}
|
|
}
|
|
|
|
/*! \brief Function called when a request is received on the session */
|
|
static int 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;
|
|
|
|
if (session->channel) {
|
|
return 0;
|
|
}
|
|
|
|
datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
|
|
if (!datastore) {
|
|
return -1;
|
|
}
|
|
|
|
transport_data = ast_calloc(1, sizeof(*transport_data));
|
|
if (!transport_data) {
|
|
return -1;
|
|
}
|
|
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) {
|
|
ast_sip_session_send_response(session, packet);
|
|
}
|
|
|
|
ast_log(LOG_ERROR, "Failed to allocate new PJSIP channel on incoming SIP INVITE\n");
|
|
return -1;
|
|
}
|
|
/* channel gets created on incoming request, but we wait to call start
|
|
so other supplements have a chance to run */
|
|
return 0;
|
|
}
|
|
|
|
static int call_pickup_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
|
|
{
|
|
struct ast_features_pickup_config *pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
|
|
struct ast_channel *chan;
|
|
|
|
/* We don't care about reinvites */
|
|
if (session->inv_session->state >= PJSIP_INV_STATE_CONFIRMED) {
|
|
return 0;
|
|
}
|
|
|
|
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;
|
|
|
|
/* We don't care about reinvites */
|
|
if (session->inv_session->state >= PJSIP_INV_STATE_CONFIRMED) {
|
|
return 0;
|
|
}
|
|
|
|
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));
|
|
|
|
return (res == AST_PBX_SUCCESS) ? 0 : -1;
|
|
}
|
|
|
|
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(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);
|
|
|
|
if (!session->channel) {
|
|
return;
|
|
}
|
|
|
|
switch (status.code) {
|
|
case 180:
|
|
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:
|
|
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
|
|
break;
|
|
case 200:
|
|
ast_queue_control(session->channel, AST_CONTROL_ANSWER);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* 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 = 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);
|
|
}
|
|
|
|
static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
|
|
{
|
|
if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
|
|
if (session->endpoint->media.direct_media.enabled && session->channel) {
|
|
ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
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 media_offer_function = {
|
|
.name = "PJSIP_MEDIA_OFFER",
|
|
.read = pjsip_acf_media_offer_read,
|
|
.write = pjsip_acf_media_offer_write
|
|
};
|
|
|
|
/*!
|
|
* \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(&media_offer_function)) {
|
|
ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
|
|
goto end;
|
|
}
|
|
|
|
if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
|
|
ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
|
|
goto end;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
if (ast_sip_session_register_supplement(&call_pickup_supplement)) {
|
|
ast_log(LOG_ERROR, "Unable to register PJSIP call pickup supplement\n");
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
|
|
goto end;
|
|
}
|
|
|
|
if (ast_sip_session_register_supplement(&pbx_start_supplement)) {
|
|
ast_log(LOG_ERROR, "Unable to register PJSIP pbx start supplement\n");
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
|
|
ast_sip_session_unregister_supplement(&call_pickup_supplement);
|
|
goto end;
|
|
}
|
|
|
|
if (ast_sip_session_register_supplement(&chan_pjsip_ack_supplement)) {
|
|
ast_log(LOG_ERROR, "Unable to register PJSIP ACK supplement\n");
|
|
ast_sip_session_unregister_supplement(&pbx_start_supplement);
|
|
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
|
|
ast_sip_session_unregister_supplement(&call_pickup_supplement);
|
|
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_custom_function_unregister(&media_offer_function);
|
|
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
|
|
ast_channel_unregister(&chan_pjsip_tech);
|
|
ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
|
|
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
}
|
|
|
|
/*! \brief Reload module */
|
|
static int reload(void)
|
|
{
|
|
return -1;
|
|
}
|
|
|
|
/*! \brief Unload the PJSIP channel from Asterisk */
|
|
static int unload_module(void)
|
|
{
|
|
ao2_cleanup(pjsip_uids_onhold);
|
|
pjsip_uids_onhold = NULL;
|
|
|
|
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(&call_pickup_supplement);
|
|
|
|
ast_custom_function_unregister(&media_offer_function);
|
|
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
|
|
|
|
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,
|
|
.reload = reload,
|
|
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
|
|
);
|