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asterisk/channels/chan_pjsip.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \author Joshua Colp <jcolp@digium.com>
*
* \brief PSJIP SIP Channel Driver
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<depend>pjproject</depend>
<depend>res_pjsip</depend>
<depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjlib.h>
ASTERISK_REGISTER_FILE()
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/file.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/causes.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/dsp.h"
#include "asterisk/stasis_endpoints.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/indications.h"
#include "asterisk/format_cache.h"
#include "asterisk/translate.h"
#include "asterisk/threadstorage.h"
#include "asterisk/features_config.h"
#include "asterisk/pickup.h"
#include "asterisk/test.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "pjsip/include/chan_pjsip.h"
#include "pjsip/include/dialplan_functions.h"
#include "pjsip/include/cli_functions.h"
AST_THREADSTORAGE(uniqueid_threadbuf);
#define UNIQUEID_BUFSIZE 256
static const char channel_type[] = "PJSIP";
static unsigned int chan_idx;
static void chan_pjsip_pvt_dtor(void *obj)
{
struct chan_pjsip_pvt *pvt = obj;
int i;
for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
ao2_cleanup(pvt->media[i]);
pvt->media[i] = NULL;
}
}
/* \brief Asterisk core interaction functions */
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);
static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text);
static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit);
static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout);
static int chan_pjsip_hangup(struct ast_channel *ast);
static int chan_pjsip_answer(struct ast_channel *ast);
static struct ast_frame *chan_pjsip_read(struct ast_channel *ast);
static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f);
static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
static int chan_pjsip_transfer(struct ast_channel *ast, const char *target);
static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int chan_pjsip_devicestate(const char *data);
static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
static const char *chan_pjsip_get_uniqueid(struct ast_channel *ast);
/*! \brief PBX interface structure for channel registration */
struct ast_channel_tech chan_pjsip_tech = {
.type = channel_type,
.description = "PJSIP Channel Driver",
.requester = chan_pjsip_request,
.send_text = chan_pjsip_sendtext,
.send_digit_begin = chan_pjsip_digit_begin,
.send_digit_end = chan_pjsip_digit_end,
.call = chan_pjsip_call,
.hangup = chan_pjsip_hangup,
.answer = chan_pjsip_answer,
.read = chan_pjsip_read,
.write = chan_pjsip_write,
.write_video = chan_pjsip_write,
.exception = chan_pjsip_read,
.indicate = chan_pjsip_indicate,
.transfer = chan_pjsip_transfer,
.fixup = chan_pjsip_fixup,
.devicestate = chan_pjsip_devicestate,
.queryoption = chan_pjsip_queryoption,
.func_channel_read = pjsip_acf_channel_read,
.get_pvt_uniqueid = chan_pjsip_get_uniqueid,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};
/*! \brief SIP session interaction functions */
static void chan_pjsip_session_begin(struct ast_sip_session *session);
static void chan_pjsip_session_end(struct ast_sip_session *session);
static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
/*! \brief SIP session supplement structure */
static struct ast_sip_session_supplement chan_pjsip_supplement = {
.method = "INVITE",
.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
.session_begin = chan_pjsip_session_begin,
.session_end = chan_pjsip_session_end,
.incoming_request = chan_pjsip_incoming_request,
.incoming_response = chan_pjsip_incoming_response,
/* It is important that this supplement runs after media has been negotiated */
.response_priority = AST_SIP_SESSION_AFTER_MEDIA,
};
static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
.method = "ACK",
.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
.incoming_request = chan_pjsip_incoming_ack,
};
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct chan_pjsip_pvt *pvt;
struct ast_sip_endpoint *endpoint;
struct ast_datastore *datastore;
if (!channel || !channel->session || !(pvt = channel->pvt) || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
return AST_RTP_GLUE_RESULT_FORBID;
}
datastore = ast_sip_session_get_datastore(channel->session, "t38");
if (datastore) {
ao2_ref(datastore, -1);
return AST_RTP_GLUE_RESULT_FORBID;
}
endpoint = channel->session->endpoint;
*instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
ao2_ref(*instance, +1);
ast_assert(endpoint != NULL);
if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
return AST_RTP_GLUE_RESULT_FORBID;
}
if (endpoint->media.direct_media.enabled) {
return AST_RTP_GLUE_RESULT_REMOTE;
}
return AST_RTP_GLUE_RESULT_LOCAL;
}
/*! \brief Function called by RTP engine to get local video RTP peer */
static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_endpoint *endpoint;
if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
return AST_RTP_GLUE_RESULT_FORBID;
}
endpoint = channel->session->endpoint;
*instance = pvt->media[SIP_MEDIA_VIDEO]->rtp;
ao2_ref(*instance, +1);
ast_assert(endpoint != NULL);
if (endpoint->media.rtp.encryption != AST_SIP_MEDIA_ENCRYPT_NONE) {
return AST_RTP_GLUE_RESULT_FORBID;
}
return AST_RTP_GLUE_RESULT_LOCAL;
}
/*! \brief Function called by RTP engine to get peer capabilities */
static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
ast_format_cap_append_from_cap(result, channel->session->endpoint->media.codecs, AST_MEDIA_TYPE_UNKNOWN);
}
/*! \brief Destructor function for \ref transport_info_data */
static void transport_info_destroy(void *obj)
{
struct transport_info_data *data = obj;
ast_free(data);
}
/*! \brief Datastore used to store local/remote addresses for the
* INVITE request that created the PJSIP channel */
static struct ast_datastore_info transport_info = {
.type = "chan_pjsip_transport_info",
.destroy = transport_info_destroy,
};
static struct ast_datastore_info direct_media_mitigation_info = { };
static int direct_media_mitigate_glare(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 0;
}
datastore = ast_sip_session_get_datastore(session, "direct_media_glare_mitigation");
if (!datastore) {
return 0;
}
/* Removing the datastore ensures we won't try to mitigate glare on subsequent reinvites */
ast_sip_session_remove_datastore(session, "direct_media_glare_mitigation");
if ((session->endpoint->media.direct_media.glare_mitigation ==
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_OUTGOING &&
session->inv_session->role == PJSIP_ROLE_UAC) ||
(session->endpoint->media.direct_media.glare_mitigation ==
AST_SIP_DIRECT_MEDIA_GLARE_MITIGATION_INCOMING &&
session->inv_session->role == PJSIP_ROLE_UAS)) {
return 1;
}
return 0;
}
/*!
* \pre chan is locked
*/
static int check_for_rtp_changes(struct ast_channel *chan, struct ast_rtp_instance *rtp,
struct ast_sip_session_media *media, int rtcp_fd)
{
int changed = 0;
if (rtp) {
changed = ast_rtp_instance_get_and_cmp_remote_address(rtp, &media->direct_media_addr);
if (media->rtp) {
ast_channel_set_fd(chan, rtcp_fd, -1);
ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 0);
}
} else if (!ast_sockaddr_isnull(&media->direct_media_addr)){
ast_sockaddr_setnull(&media->direct_media_addr);
changed = 1;
if (media->rtp) {
ast_rtp_instance_set_prop(media->rtp, AST_RTP_PROPERTY_RTCP, 1);
ast_channel_set_fd(chan, rtcp_fd, ast_rtp_instance_fd(media->rtp, 1));
}
}
return changed;
}
struct rtp_direct_media_data {
struct ast_channel *chan;
struct ast_rtp_instance *rtp;
struct ast_rtp_instance *vrtp;
struct ast_format_cap *cap;
struct ast_sip_session *session;
};
static void rtp_direct_media_data_destroy(void *data)
{
struct rtp_direct_media_data *cdata = data;
ao2_cleanup(cdata->session);
ao2_cleanup(cdata->cap);
ao2_cleanup(cdata->vrtp);
ao2_cleanup(cdata->rtp);
ao2_cleanup(cdata->chan);
}
static struct rtp_direct_media_data *rtp_direct_media_data_create(
struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp,
const struct ast_format_cap *cap, struct ast_sip_session *session)
{
struct rtp_direct_media_data *cdata = ao2_alloc(sizeof(*cdata), rtp_direct_media_data_destroy);
if (!cdata) {
return NULL;
}
cdata->chan = ao2_bump(chan);
cdata->rtp = ao2_bump(rtp);
cdata->vrtp = ao2_bump(vrtp);
cdata->cap = ao2_bump((struct ast_format_cap *)cap);
cdata->session = ao2_bump(session);
return cdata;
}
static int send_direct_media_request(void *data)
{
struct rtp_direct_media_data *cdata = data;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(cdata->chan);
struct chan_pjsip_pvt *pvt = channel->pvt;
int changed = 0;
int res = 0;
/* The channel needs to be locked when checking for RTP changes.
* Otherwise, we could end up destroying an underlying RTCP structure
* at the same time that the channel thread is attempting to read RTCP
*/
ast_channel_lock(cdata->chan);
if (pvt->media[SIP_MEDIA_AUDIO]) {
changed |= check_for_rtp_changes(
cdata->chan, cdata->rtp, pvt->media[SIP_MEDIA_AUDIO], 1);
}
if (pvt->media[SIP_MEDIA_VIDEO]) {
changed |= check_for_rtp_changes(
cdata->chan, cdata->vrtp, pvt->media[SIP_MEDIA_VIDEO], 3);
}
ast_channel_unlock(cdata->chan);
if (direct_media_mitigate_glare(cdata->session)) {
ast_debug(4, "Disregarding setting RTP on %s: mitigating re-INVITE glare\n", ast_channel_name(cdata->chan));
ao2_ref(cdata, -1);
return 0;
}
if (cdata->cap && ast_format_cap_count(cdata->cap) &&
!ast_format_cap_identical(cdata->session->direct_media_cap, cdata->cap)) {
ast_format_cap_remove_by_type(cdata->session->direct_media_cap, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_append_from_cap(cdata->session->direct_media_cap, cdata->cap, AST_MEDIA_TYPE_UNKNOWN);
changed = 1;
}
if (changed) {
ast_debug(4, "RTP changed on %s; initiating direct media update\n", ast_channel_name(cdata->chan));
res = ast_sip_session_refresh(cdata->session, NULL, NULL, NULL,
cdata->session->endpoint->media.direct_media.method, 1);
}
ao2_ref(cdata, -1);
return res;
}
/*! \brief Function called by RTP engine to change where the remote party should send media */
static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
struct ast_rtp_instance *rtp,
struct ast_rtp_instance *vrtp,
struct ast_rtp_instance *tpeer,
const struct ast_format_cap *cap,
int nat_active)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct ast_sip_session *session = channel->session;
struct rtp_direct_media_data *cdata;
/* Don't try to do any direct media shenanigans on early bridges */
if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) {
ast_debug(4, "Disregarding setting RTP on %s: channel is not bridged\n", ast_channel_name(chan));
return 0;
}
if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
ast_debug(4, "Disregarding setting RTP on %s: NAT is active\n", ast_channel_name(chan));
return 0;
}
cdata = rtp_direct_media_data_create(chan, rtp, vrtp, cap, session);
if (!cdata) {
return 0;
}
if (ast_sip_push_task(session->serializer, send_direct_media_request, cdata)) {
ast_log(LOG_ERROR, "Unable to send direct media request for channel %s\n", ast_channel_name(chan));
ao2_ref(cdata, -1);
}
return 0;
}
/*! \brief Local glue for interacting with the RTP engine core */
static struct ast_rtp_glue chan_pjsip_rtp_glue = {
.type = "PJSIP",
.get_rtp_info = chan_pjsip_get_rtp_peer,
.get_vrtp_info = chan_pjsip_get_vrtp_peer,
.get_codec = chan_pjsip_get_codec,
.update_peer = chan_pjsip_set_rtp_peer,
};
static void set_channel_on_rtp_instance(struct chan_pjsip_pvt *pvt, const char *channel_id)
{
if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, channel_id);
}
if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, channel_id);
}
}
/*! \brief Function called to create a new PJSIP Asterisk channel */
static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *cid_name)
{
struct ast_channel *chan;
struct ast_format_cap *caps;
RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
struct ast_sip_channel_pvt *channel;
struct ast_variable *var;
if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
return NULL;
}
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!caps) {
return NULL;
}
chan = ast_channel_alloc_with_endpoint(1, state,
S_COR(session->id.number.valid, session->id.number.str, ""),
S_COR(session->id.name.valid, session->id.name.str, ""),
session->endpoint->accountcode,
exten, session->endpoint->context,
assignedids, requestor, 0,
session->endpoint->persistent, "PJSIP/%s-%08x",
ast_sorcery_object_get_id(session->endpoint),
(unsigned) ast_atomic_fetchadd_int((int *) &chan_idx, +1));
if (!chan) {
ao2_ref(caps, -1);
return NULL;
}
ast_channel_tech_set(chan, &chan_pjsip_tech);
if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
ao2_ref(caps, -1);
ast_channel_unlock(chan);
ast_hangup(chan);
return NULL;
}
ast_channel_stage_snapshot(chan);
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);
if (!ast_format_cap_empty(caps)) {
struct ast_format *fmt;
fmt = ast_format_cap_get_best_by_type(caps, AST_MEDIA_TYPE_AUDIO);
if (!fmt) {
/* Since our capabilities aren't empty, this will succeed */
fmt = ast_format_cap_get_format(caps, 0);
}
ast_channel_set_writeformat(chan, fmt);
ast_channel_set_rawwriteformat(chan, fmt);
ast_channel_set_readformat(chan, fmt);
ast_channel_set_rawreadformat(chan, fmt);
ao2_ref(fmt, -1);
}
ao2_ref(caps, -1);
if (state == AST_STATE_RING) {
ast_channel_rings_set(chan, 1);
}
ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
ast_party_id_copy(&ast_channel_caller(chan)->id, &session->id);
ast_party_id_copy(&ast_channel_caller(chan)->ani, &session->id);
ast_channel_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;
if (session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
return 0;
}
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);
}
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);
struct ast_sip_session *session;
if (ast_channel_state(ast) == AST_STATE_UP) {
return 0;
}
ast_setstate(ast, AST_STATE_UP);
session = ao2_bump(channel->session);
/* the answer task needs to be pushed synchronously otherwise a race condition
can occur between this thread and bridging (specifically when native bridging
attempts to do direct media) */
ast_channel_unlock(ast);
if (ast_sip_push_task_synchronous(session->serializer, answer, session)) {
ast_log(LOG_WARNING, "Unable to push answer task to the threadpool. Cannot answer call\n");
ao2_ref(session, -1);
ast_channel_lock(ast);
return -1;
}
ao2_ref(session, -1);
ast_channel_lock(ast);
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;
int dsp_features;
dsp_features = ast_dsp_get_features(session->dsp);
dsp_features &= ~DSP_FEATURE_FAX_DETECT;
if (dsp_features) {
ast_dsp_set_features(session->dsp, dsp_features);
} else {
ast_dsp_free(session->dsp);
session->dsp = NULL;
}
/* If already executing in the fax extension don't do anything */
if (!strcmp(ast_channel_exten(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 ast_sip_session *session;
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;
}
ast_rtp_instance_set_last_rx(media->rtp, time(NULL));
if (f->frametype != AST_FRAME_VOICE) {
return f;
}
session = channel->session;
if (ast_format_cap_iscompatible_format(session->endpoint->media.codecs, f->subclass.format) == AST_FORMAT_CMP_NOT_EQUAL) {
ast_debug(1, "Oooh, got a frame with format of %s on channel '%s' when endpoint '%s' is not configured for it\n",
ast_format_get_name(f->subclass.format), ast_channel_name(ast),
ast_sorcery_object_get_id(session->endpoint));
ast_frfree(f);
return &ast_null_frame;
}
if (session->dsp) {
int dsp_features;
dsp_features = ast_dsp_get_features(session->dsp);
if ((dsp_features & DSP_FEATURE_FAX_DETECT)
&& session->endpoint->faxdetect_timeout
&& session->endpoint->faxdetect_timeout <= ast_channel_get_up_time(ast)) {
dsp_features &= ~DSP_FEATURE_FAX_DETECT;
if (dsp_features) {
ast_dsp_set_features(session->dsp, dsp_features);
} else {
ast_dsp_free(session->dsp);
session->dsp = NULL;
}
ast_debug(3, "Channel driver fax CNG detection timeout on %s\n",
ast_channel_name(ast));
}
}
if (session->dsp) {
f = ast_dsp_process(ast, session->dsp, f);
if (f && (f->frametype == AST_FRAME_DTMF)) {
if (f->subclass.integer == 'f') {
ast_debug(3, "Channel driver fax CNG detected on %s\n",
ast_channel_name(ast));
f = chan_pjsip_cng_tone_detected(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(AST_FORMAT_CAP_NAMES_LEN);
struct ast_str *write_transpath = ast_str_alloca(256);
struct ast_str *read_transpath = ast_str_alloca(256);
ast_log(LOG_WARNING,
"Channel %s asked to send %s frame when native formats are %s (rd:%s->%s;%s wr:%s->%s;%s)\n",
ast_channel_name(ast),
ast_format_get_name(frame->subclass.format),
ast_format_cap_get_names(ast_channel_nativeformats(ast), &cap_buf),
ast_format_get_name(ast_channel_rawreadformat(ast)),
ast_format_get_name(ast_channel_readformat(ast)),
ast_translate_path_to_str(ast_channel_readtrans(ast), &read_transpath),
ast_format_get_name(ast_channel_writeformat(ast)),
ast_format_get_name(ast_channel_rawwriteformat(ast)),
ast_translate_path_to_str(ast_channel_writetrans(ast), &write_transpath));
return 0;
}
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;
}
/*! \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 chan_pjsip_pvt *pvt = channel->pvt;
if (channel->session->channel != oldchan) {
return -1;
}
/*
* The masquerade has suspended the channel's session
* serializer so we can safely change it outside of
* the serializer thread.
*/
channel->session->channel = newchan;
set_channel_on_rtp_instance(pvt, ast_channel_uniqueid(newchan));
return 0;
}
/*! AO2 hash function for on hold UIDs */
static int uid_hold_hash_fn(const void *obj, const int flags)
{
const char *key = obj;
switch (flags & OBJ_SEARCH_MASK) {
case OBJ_SEARCH_KEY:
break;
case OBJ_SEARCH_OBJECT:
break;
default:
/* Hash can only work on something with a full key. */
ast_assert(0);
return 0;
}
return ast_str_hash(key);
}
/*! AO2 sort function for on hold UIDs */
static int uid_hold_sort_fn(const void *obj_left, const void *obj_right, const int flags)
{
const char *left = obj_left;
const char *right = obj_right;
int cmp;
switch (flags & OBJ_SEARCH_MASK) {
case OBJ_SEARCH_OBJECT:
case OBJ_SEARCH_KEY:
cmp = strcmp(left, right);
break;
case OBJ_SEARCH_PARTIAL_KEY:
cmp = strncmp(left, right, strlen(right));
break;
default:
/* Sort can only work on something with a full or partial key. */
ast_assert(0);
cmp = 0;
break;
}
return cmp;
}
static struct ao2_container *pjsip_uids_onhold;
/*!
* \brief Add a channel ID to the list of PJSIP channels on hold
*
* \param chan_uid - Unique ID of the channel being put into the hold list
*
* \retval 0 Channel has been added to or was already in the hold list
* \retval -1 Failed to add channel to the hold list
*/
static int chan_pjsip_add_hold(const char *chan_uid)
{
RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
if (hold_uid) {
/* Device is already on hold. Nothing to do. */
return 0;
}
/* Device wasn't in hold list already. Create a new one. */
hold_uid = ao2_alloc_options(strlen(chan_uid) + 1, NULL,
AO2_ALLOC_OPT_LOCK_NOLOCK);
if (!hold_uid) {
return -1;
}
ast_copy_string(hold_uid, chan_uid, strlen(chan_uid) + 1);
if (ao2_link(pjsip_uids_onhold, hold_uid) == 0) {
return -1;
}
return 0;
}
/*!
* \brief Remove a channel ID from the list of PJSIP channels on hold
*
* \param chan_uid - Unique ID of the channel being taken out of the hold list
*/
static void chan_pjsip_remove_hold(const char *chan_uid)
{
ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY | OBJ_UNLINK | OBJ_NODATA);
}
/*!
* \brief Determine whether a channel ID is in the list of PJSIP channels on hold
*
* \param chan_uid - Channel being checked
*
* \retval 0 The channel is not in the hold list
* \retval 1 The channel is in the hold list
*/
static int chan_pjsip_get_hold(const char *chan_uid)
{
RAII_VAR(char *, hold_uid, NULL, ao2_cleanup);
hold_uid = ao2_find(pjsip_uids_onhold, chan_uid, OBJ_SEARCH_KEY);
if (!hold_uid) {
return 0;
}
return 1;
}
/*! \brief Function called to get the device state of an endpoint */
static int chan_pjsip_devicestate(const char *data)
{
RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
enum ast_device_state state = AST_DEVICE_UNKNOWN;
RAII_VAR(struct ast_endpoint_snapshot *, endpoint_snapshot, NULL, ao2_cleanup);
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_t38_state 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 ((session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
(pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS)) {
ast_sip_session_send_response(session, packet);
}
ao2_ref(ind_data, -1);
return 0;
}
/*! \brief Send SIP INFO with video update request */
static int transmit_info_with_vidupdate(void *data)
{
const char * xml =
"<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
" <media_control>\r\n"
" <vc_primitive>\r\n"
" <to_encoder>\r\n"
" <picture_fast_update/>\r\n"
" </to_encoder>\r\n"
" </vc_primitive>\r\n"
" </media_control>\r\n";
const struct ast_sip_body body = {
.type = "application",
.subtype = "media_control+xml",
.body_text = xml
};
RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
struct pjsip_tx_data *tdata;
if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) {
ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
return -1;
}
if (ast_sip_add_body(tdata, &body)) {
ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
return -1;
}
ast_sip_session_send_request(session, tdata);
return 0;
}
/*!
* \internal
* \brief TRUE if a COLP update can be sent to the peer.
* \since 13.3.0
*
* \param session The session to see if the COLP update is allowed.
*
* \retval 0 Update is not allowed.
* \retval 1 Update is allowed.
*/
static int is_colp_update_allowed(struct ast_sip_session *session)
{
struct ast_party_id connected_id;
int update_allowed = 0;
if (!session->endpoint->id.send_pai && !session->endpoint->id.send_rpid) {
return 0;
}
/*
* Check if privacy allows the update. Check while the channel
* is locked so we can work with the shallow connected_id copy.
*/
ast_channel_lock(session->channel);
connected_id = ast_channel_connected_effective_id(session->channel);
if (connected_id.number.valid
&& (session->endpoint->id.trust_outbound
|| (ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED)) {
update_allowed = 1;
}
ast_channel_unlock(session->channel);
return update_allowed;
}
/*! \brief Update connected line information */
static int update_connected_line_information(void *data)
{
struct ast_sip_session *session = data;
if (ast_channel_state(session->channel) == AST_STATE_UP
|| session->inv_session->role == PJSIP_ROLE_UAC) {
if (is_colp_update_allowed(session)) {
enum ast_sip_session_refresh_method method;
int generate_new_sdp;
method = session->endpoint->id.refresh_method;
if (session->inv_session->invite_tsx
&& (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
}
/* Only the INVITE method actually needs SDP, UPDATE can do without */
generate_new_sdp = (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE);
ast_sip_session_refresh(session, NULL, NULL, NULL, method, generate_new_sdp);
}
} else if (session->endpoint->id.rpid_immediate
&& session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED
&& is_colp_update_allowed(session)) {
int response_code = 0;
if (ast_channel_state(session->channel) == AST_STATE_RING) {
response_code = !session->endpoint->inband_progress ? 180 : 183;
} else if (ast_channel_state(session->channel) == AST_STATE_RINGING) {
response_code = 183;
}
if (response_code) {
struct pjsip_tx_data *packet = NULL;
if (pjsip_inv_answer(session->inv_session, response_code, NULL, NULL, &packet) == PJ_SUCCESS) {
ast_sip_session_send_response(session, packet);
}
}
}
ao2_ref(session, -1);
return 0;
}
/*! \brief Callback which changes the value of locally held on the media stream */
static int local_hold_set_state(void *obj, void *arg, int flags)
{
struct ast_sip_session_media *session_media = obj;
unsigned int *held = arg;
session_media->locally_held = *held;
return 0;
}
/*! \brief Update local hold state and send a re-INVITE with the new SDP */
static int remote_send_hold_refresh(struct ast_sip_session *session, unsigned int held)
{
ao2_callback(session->media, OBJ_NODATA, local_hold_set_state, &held);
ast_sip_session_refresh(session, NULL, NULL, NULL, AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1);
ao2_ref(session, -1);
return 0;
}
/*! \brief Update local hold state to be held */
static int remote_send_hold(void *data)
{
return remote_send_hold_refresh(data, 1);
}
/*! \brief Update local hold state to be unheld */
static int remote_send_unhold(void *data)
{
return remote_send_hold_refresh(data, 0);
}
/*! \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_MASQUERADE_NOTIFY:
ast_assert(datalen == sizeof(int));
if (*(int *) data) {
/*
* Masquerade is beginning:
* Wait for session serializer to get suspended.
*/
ast_channel_unlock(ast);
ast_sip_session_suspend(channel->session);
ast_channel_lock(ast);
} else {
/*
* Masquerade is complete:
* Unsuspend the session serializer.
*/
ast_sip_session_unsuspend(channel->session);
}
break;
case AST_CONTROL_HOLD:
chan_pjsip_add_hold(ast_channel_uniqueid(ast));
device_buf_size = strlen(ast_channel_name(ast)) + 1;
device_buf = alloca(device_buf_size);
ast_channel_get_device_name(ast, device_buf, device_buf_size);
ast_devstate_changed_literal(AST_DEVICE_ONHOLD, 1, device_buf);
if (!channel->session->endpoint->moh_passthrough) {
ast_moh_start(ast, data, NULL);
} else {
if (ast_sip_push_task(channel->session->serializer, remote_send_hold, ao2_bump(channel->session))) {
ast_log(LOG_WARNING, "Could not queue task to remotely put session '%s' on hold with endpoint '%s'\n",
ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
ao2_ref(channel->session, -1);
}
}
break;
case AST_CONTROL_UNHOLD:
chan_pjsip_remove_hold(ast_channel_uniqueid(ast));
device_buf_size = strlen(ast_channel_name(ast)) + 1;
device_buf = alloca(device_buf_size);
ast_channel_get_device_name(ast, device_buf, device_buf_size);
ast_devstate_changed_literal(AST_DEVICE_UNKNOWN, 1, device_buf);
if (!channel->session->endpoint->moh_passthrough) {
ast_moh_stop(ast);
} else {
if (ast_sip_push_task(channel->session->serializer, remote_send_unhold, ao2_bump(channel->session))) {
ast_log(LOG_WARNING, "Could not queue task to remotely take session '%s' off hold with endpoint '%s'\n",
ast_sorcery_object_get_id(channel->session), ast_sorcery_object_get_id(channel->session->endpoint));
ao2_ref(channel->session, -1);
}
}
break;
case AST_CONTROL_SRCUPDATE:
break;
case AST_CONTROL_SRCCHANGE:
break;
case AST_CONTROL_REDIRECTING:
if (ast_channel_state(ast) != AST_STATE_UP) {
response_code = 181;
} else {
res = -1;
}
break;
case AST_CONTROL_T38_PARAMETERS:
res = 0;
if (channel->session->t38state == T38_PEER_REINVITE) {
const struct ast_control_t38_parameters *parameters = data;
if (parameters->request_response == AST_T38_REQUEST_PARMS) {
res = AST_T38_REQUEST_PARMS;
}
}
break;
case -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
|| !packet) {
ast_log(LOG_WARNING, "Failed to redirect PJSIP session for channel %s\n",
ast_channel_name(session->channel));
message = AST_TRANSFER_FAILED;
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
return;
}
if (!(contact = pjsip_msg_find_hdr(packet->msg, PJSIP_H_CONTACT, NULL))) {
contact = pjsip_contact_hdr_create(packet->pool);
}
pj_strdup2_with_null(packet->pool, &tmp, target);
if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
ast_log(LOG_WARNING, "Failed to parse destination URI '%s' for channel %s\n",
target, ast_channel_name(session->channel));
message = AST_TRANSFER_FAILED;
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
pjsip_tx_data_dec_ref(packet);
return;
}
pjsip_msg_add_hdr(packet->msg, (pjsip_hdr *) contact);
ast_sip_session_send_response(session, packet);
ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
static void transfer_refer(struct ast_sip_session *session, const char *target)
{
pjsip_evsub *sub;
enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
pj_str_t tmp;
pjsip_tx_data *packet;
const char *ref_by_val;
char local_info[pj_strlen(&session->inv_session->dlg->local.info_str) + 1];
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;
}
ref_by_val = pbx_builtin_getvar_helper(session->channel, "SIPREFERREDBYHDR");
if (!ast_strlen_zero(ref_by_val)) {
ast_sip_add_header(packet, "Referred-By", ref_by_val);
} else {
ast_copy_pj_str(local_info, &session->inv_session->dlg->local.info_str, sizeof(local_info));
ast_sip_add_header(packet, "Referred-By", local_info);
}
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;
struct ast_sip_endpoint *endpoint = NULL;
struct ast_sip_contact *contact = NULL;
const char *target = trnf_data->target;
/* See if we have an endpoint; if so, use its contact */
endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", target);
if (endpoint) {
contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
if (contact && !ast_strlen_zero(contact->uri)) {
target = contact->uri;
}
}
if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
transfer_redirect(trnf_data->session, target);
} else {
transfer_refer(trnf_data->session, target);
}
ao2_ref(trnf_data, -1);
ao2_cleanup(endpoint);
ao2_cleanup(contact);
return 0;
}
/*! \brief Function called by core for Asterisk initiated transfer */
static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
if (!trnf_data) {
return -1;
}
if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
ast_log(LOG_WARNING, "Error requesting transfer\n");
ao2_cleanup(trnf_data);
return -1;
}
return 0;
}
/*! \brief Function called by core to start a DTMF digit */
static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct 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);
break;
case AST_SIP_DTMF_AUTO:
if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
return -1;
}
ast_rtp_instance_dtmf_begin(media->rtp, digit);
break;
case AST_SIP_DTMF_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);
break;
case AST_SIP_DTMF_AUTO:
if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
return -1;
}
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
break;
case AST_SIP_DTMF_NONE:
break;
case AST_SIP_DTMF_INBAND:
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;
ast_sip_session_terminate(session, cause);
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;
int cause;
struct hangup_data *h_data;
if (!channel || !channel->session) {
return -1;
}
pvt = channel->pvt;
cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
h_data = hangup_data_alloc(cause, ast);
if (!h_data) {
goto failure;
}
if (ast_sip_push_task(channel->session->serializer, hangup, h_data)) {
ast_log(LOG_WARNING, "Unable to push hangup task to the threadpool. Expect bad things\n");
goto failure;
}
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;
return -1;
} 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))) {
ast_log(LOG_ERROR, "Failed to create outgoing session to endpoint '%s'\n", endpoint_name);
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
};
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;
}
/* Check for a to-tag to determine if this is a reinvite */
if (rdata->msg_info.to->tag.slen) {
/* Weird case. We've received a reinvite but we don't have a channel. The most
* typical case for this happening is that a blind transfer fails, and so the
* transferer attempts to reinvite himself back into the call. We already got
* rid of that channel, and the other side of the call is unrecoverable.
*
* We treat this as a failure, so our best bet is to just hang this call
* up and not create a new channel. Clearing defer_terminate here ensures that
* calling ast_sip_session_terminate() can result in a BYE being sent ASAP.
*/
session->defer_terminate = 0;
ast_sip_session_terminate(session, 400);
return -1;
}
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
&& packet) {
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;
struct ast_channel *chan;
/* Check for a to-tag to determine if this is a reinvite */
if (rdata->msg_info.to->tag.slen) {
/* We don't care about reinvites */
return 0;
}
pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
if (!pickup_cfg) {
ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension.\n");
return 0;
}
if (strcmp(session->exten, pickup_cfg->pickupexten)) {
ao2_ref(pickup_cfg, -1);
return 0;
}
ao2_ref(pickup_cfg, -1);
/* We can't directly use session->channel because the pickup operation will cause a masquerade to occur,
* changing the channel pointer in session to a different channel. To ensure we work on the right channel
* we store a pointer locally before we begin and keep a reference so it remains valid no matter what.
*/
chan = ast_channel_ref(session->channel);
if (ast_pickup_call(chan)) {
ast_channel_hangupcause_set(chan, AST_CAUSE_CALL_REJECTED);
} else {
ast_channel_hangupcause_set(chan, AST_CAUSE_NORMAL_CLEARING);
}
/* A hangup always occurs because the pickup operation will have either failed resulting in the call
* needing to be hung up OR the pickup operation was a success and the channel we now have is actually
* the channel that was replaced, which should be hung up since it is literally in limbo not connected
* to anything at all.
*/
ast_hangup(chan);
ast_channel_unref(chan);
return 1;
}
static struct ast_sip_session_supplement call_pickup_supplement = {
.method = "INVITE",
.priority = AST_SIP_SUPPLEMENT_PRIORITY_LAST - 1,
.incoming_request = call_pickup_incoming_request,
};
static int pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
{
int res;
/* Check for a to-tag to determine if this is a reinvite */
if (rdata->msg_info.to->tag.slen) {
/* We don't care about reinvites */
return 0;
}
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;
CHECK_PJSIP_SESSION_MODULE_LOADED();
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;
}
if (pjsip_channel_cli_register()) {
ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
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 Unload the PJSIP channel from Asterisk */
static int unload_module(void)
{
ao2_cleanup(pjsip_uids_onhold);
pjsip_uids_onhold = NULL;
pjsip_channel_cli_unregister();
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
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,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
);
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