The large GULP->PJSIP renaming effort.

The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:

* The word "Gulp" in dialstrings, functions, and CLI commands is now
  "PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 2107 insertions, 0 deletions

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
new file mode 100644
index 000000000..527ba2793
--- /dev/null
+++ b/channels/chan_pjsip.c
@@ -0,0 +1,2107 @@
+/*
+ * 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_FILE_VERSION(__FILE__, "$Revision$")
+
+#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/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+
+/*** DOCUMENTATION
+	<function name="PJSIP_DIAL_CONTACTS" language="en_US">
+		<synopsis>
+			Return a dial string for dialing all contacts on an AOR.
+		</synopsis>
+		<syntax>
+			<parameter name="endpoint" required="true">
+				<para>Name of the endpoint</para>
+			</parameter>
+			<parameter name="aor" required="false">
+				<para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
+			</parameter>
+			<parameter name="request_user" required="false">
+				<para>Optional request user to use in the request URI</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
+		</description>
+	</function>
+	<function name="PJSIP_MEDIA_OFFER" language="en_US">
+		<synopsis>
+			Media and codec offerings to be set on an outbound SIP channel prior to dialing.
+		</synopsis>
+		<syntax>
+			<parameter name="media" required="true">
+				<para>types of media offered</para>
+			</parameter>
+		</syntax>
+		<description>
+			<para>Returns the codecs offered based upon the media choice</para>
+		</description>
+	</function>
+ ***/
+
+static const char desc[] = "PJSIP Channel";
+static const char channel_type[] = "PJSIP";
+
+static unsigned int chan_idx;
+
+/*!
+ * \brief Positions of various media
+ */
+enum sip_session_media_position {
+	/*! \brief First is audio */
+	SIP_MEDIA_AUDIO = 0,
+	/*! \brief Second is video */
+	SIP_MEDIA_VIDEO,
+	/*! \brief Last is the size for media details */
+	SIP_MEDIA_SIZE,
+};
+
+struct chan_pjsip_pvt {
+	struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
+};
+
+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_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);
+
+/*! \brief PBX interface structure for channel registration */
+static 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,
+	.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_SESSION_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,
+};
+
+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_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
+	.incoming_request = chan_pjsip_incoming_ack,
+};
+
+/*! \brief Dialplan function for constructing a dial string for calling all contacts */
+static int chan_pjsip_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+	RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
+	RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
+	const char *aor_name;
+	char *rest;
+
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(endpoint_name);
+		AST_APP_ARG(aor_name);
+		AST_APP_ARG(request_user);
+	);
+
+	AST_STANDARD_APP_ARGS(args, data);
+
+	if (ast_strlen_zero(args.endpoint_name)) {
+		ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
+		return -1;
+	} else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
+		ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
+		return -1;
+	}
+
+	aor_name = S_OR(args.aor_name, endpoint->aors);
+
+	if (ast_strlen_zero(aor_name)) {
+		ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
+		return -1;
+	} else if (!(dial = ast_str_create(len))) {
+		ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
+		return -1;
+	} else if (!(rest = ast_strdupa(aor_name))) {
+		ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
+		return -1;
+	}
+
+	while ((aor_name = strsep(&rest, ","))) {
+		RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
+		RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
+		struct ao2_iterator it_contacts;
+		struct ast_sip_contact *contact;
+
+		if (!aor) {
+			/* If the AOR provided is not found skip it, there may be more */
+			continue;
+		} else if (!(contacts = ast_sip_location_retrieve_aor_contacts(aor))) {
+			/* No contacts are available, skip it as well */
+			continue;
+		} else if (!ao2_container_count(contacts)) {
+			/* We were given a container but no contacts are in it... */
+			continue;
+		}
+
+		it_contacts = ao2_iterator_init(contacts, 0);
+		for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
+			ast_str_append(&dial, -1, "PJSIP/");
+
+			if (!ast_strlen_zero(args.request_user)) {
+				ast_str_append(&dial, -1, "%s@", args.request_user);
+			}
+			ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
+		}
+		ao2_iterator_destroy(&it_contacts);
+	}
+
+	/* Trim the '&' at the end off */
+	ast_str_truncate(dial, ast_str_strlen(dial) - 1);
+
+	ast_copy_string(buf, ast_str_buffer(dial), len);
+
+	return 0;
+}
+
+static struct ast_custom_function chan_pjsip_dial_contacts_function = {
+	.name = "PJSIP_DIAL_CONTACTS",
+	.read = chan_pjsip_dial_contacts,
+};
+
+static int media_offer_read_av(struct ast_sip_session *session, char *buf,
+			       size_t len, enum ast_format_type media_type)
+{
+	int i, size = 0;
+	struct ast_format fmt;
+	const char *name;
+
+	for (i = 0; ast_codec_pref_index(&session->override_prefs, i, &fmt); ++i) {
+		if (AST_FORMAT_GET_TYPE(fmt.id) != media_type) {
+			continue;
+		}
+
+		name = ast_getformatname(&fmt);
+
+		if (ast_strlen_zero(name)) {
+			ast_log(LOG_WARNING, "PJSIP_MEDIA_OFFER unrecognized format %s\n", name);
+			continue;
+		}
+
+		/* add one since we'll include a comma */
+		size = strlen(name) + 1;
+		len -= size;
+		if ((len) < 0) {
+			break;
+		}
+
+		/* no reason to use strncat here since we have already ensured buf has
+                   enough space, so strcat can be safely used */
+		strcat(buf, name);
+		strcat(buf, ",");
+	}
+
+	if (size) {
+		/* remove the extra comma */
+		buf[strlen(buf) - 1] = '\0';
+	}
+	return 0;
+}
+
+struct media_offer_data {
+	struct ast_sip_session *session;
+	enum ast_format_type media_type;
+	const char *value;
+};
+
+static int media_offer_write_av(void *obj)
+{
+	struct media_offer_data *data = obj;
+	int i;
+	struct ast_format fmt;
+	/* remove all of the given media type first */
+	for (i = 0; ast_codec_pref_index(&data->session->override_prefs, i, &fmt); ++i) {
+		if (AST_FORMAT_GET_TYPE(fmt.id) == data->media_type) {
+			ast_codec_pref_remove(&data->session->override_prefs, &fmt);
+		}
+	}
+	ast_format_cap_remove_bytype(data->session->req_caps, data->media_type);
+	ast_parse_allow_disallow(&data->session->override_prefs, data->session->req_caps, data->value, 1);
+
+	return 0;
+}
+
+static int media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+
+	if (!strcmp(data, "audio")) {
+		return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_AUDIO);
+	} else if (!strcmp(data, "video")) {
+		return media_offer_read_av(channel->session, buf, len, AST_FORMAT_TYPE_VIDEO);
+	}
+
+	return 0;
+}
+
+static int media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
+{
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+
+	struct media_offer_data mdata = {
+		.session = channel->session,
+		.value = value
+	};
+
+	if (!strcmp(data, "audio")) {
+		mdata.media_type = AST_FORMAT_TYPE_AUDIO;
+	} else if (!strcmp(data, "video")) {
+		mdata.media_type = AST_FORMAT_TYPE_VIDEO;
+	}
+
+	return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
+}
+
+static struct ast_custom_function media_offer_function = {
+	.name = "PJSIP_MEDIA_OFFER",
+	.read = media_offer_read,
+	.write = media_offer_write
+};
+
+/*! \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 = channel->pvt;
+	struct ast_sip_endpoint *endpoint;
+
+	if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
+		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_copy(result, channel->session->endpoint->media.codecs);
+}
+
+static int send_direct_media_request(void *data)
+{
+	RAII_VAR(struct ast_sip_session *, session, data, ao2_cleanup);
+
+	return ast_sip_session_refresh(session, NULL, NULL, NULL,
+			session->endpoint->media.direct_media.method, 1);
+}
+
+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;
+}
+
+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;
+}
+
+/*! \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 chan_pjsip_pvt *pvt = channel->pvt;
+	struct ast_sip_session *session = channel->session;
+	int changed = 0;
+	struct ast_channel *bridge_peer;
+
+	/* Don't try to do any direct media shenanigans on early bridges */
+	bridge_peer = ast_channel_bridge_peer(chan);
+	if ((rtp || vrtp || tpeer) && !bridge_peer) {
+		return 0;
+	}
+	ast_channel_cleanup(bridge_peer);
+
+	if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
+		return 0;
+	}
+
+	if (pvt->media[SIP_MEDIA_AUDIO]) {
+		changed |= check_for_rtp_changes(chan, rtp, pvt->media[SIP_MEDIA_AUDIO], 1);
+	}
+	if (pvt->media[SIP_MEDIA_VIDEO]) {
+		changed |= check_for_rtp_changes(chan, vrtp, pvt->media[SIP_MEDIA_VIDEO], 3);
+	}
+
+	if (direct_media_mitigate_glare(session)) {
+		return 0;
+	}
+
+	if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(session->direct_media_cap, cap)) {
+		ast_format_cap_copy(session->direct_media_cap, cap);
+		changed = 1;
+	}
+
+	if (changed) {
+		ao2_ref(session, +1);
+
+
+		if (ast_sip_push_task(session->serializer, send_direct_media_request, session)) {
+			ao2_cleanup(session);
+		}
+	}
+
+	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,
+};
+
+/*! \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 char *linkedid, const char *cid_name)
+{
+	struct ast_channel *chan;
+	struct ast_format fmt;
+	RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
+	struct ast_sip_channel_pvt *channel;
+
+	if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
+		return NULL;
+	}
+
+	if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "PJSIP/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
+		ast_atomic_fetchadd_int((int *)&chan_idx, +1)))) {
+		return NULL;
+	}
+
+	ast_channel_tech_set(chan, &chan_pjsip_tech);
+
+	if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
+		ast_hangup(chan);
+		return NULL;
+	}
+
+	/* If res_sip_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);
+	ast_channel_tech_pvt_set(chan, channel);
+	if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
+		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(chan));
+	}
+	if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
+		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, ast_channel_uniqueid(chan));
+	}
+
+	if (ast_format_cap_is_empty(session->req_caps) || !ast_format_cap_has_joint(session->req_caps, session->endpoint->media.codecs)) {
+		ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->media.codecs);
+	} else {
+		ast_format_cap_copy(ast_channel_nativeformats(chan), session->req_caps);
+	}
+
+	ast_codec_choose(&session->endpoint->media.prefs, ast_channel_nativeformats(chan), 1, &fmt);
+	ast_format_copy(ast_channel_writeformat(chan), &fmt);
+	ast_format_copy(ast_channel_rawwriteformat(chan), &fmt);
+	ast_format_copy(ast_channel_readformat(chan), &fmt);
+	ast_format_copy(ast_channel_rawreadformat(chan), &fmt);
+
+	if (state == AST_STATE_RING) {
+		ast_channel_rings_set(chan, 1);
+	}
+
+	ast_channel_adsicpe_set(chan, AST_ADSI_UNAVAILABLE);
+
+	ast_channel_context_set(chan, session->endpoint->context);
+	ast_channel_exten_set(chan, S_OR(exten, "s"));
+	ast_channel_priority_set(chan, 1);
+
+	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);
+	}
+
+	ast_endpoint_add_channel(session->endpoint->persistent, chan);
+
+	return chan;
+}
+
+static int answer(void *data)
+{
+	pj_status_t status;
+	pjsip_tx_data *packet;
+	struct ast_sip_session *session = data;
+
+	if ((status = pjsip_inv_answer(session->inv_session, 200, NULL, NULL, &packet)) == PJ_SUCCESS) {
+		ast_sip_session_send_response(session, packet);
+	}
+
+	ao2_ref(session, -1);
+
+	return (status == PJ_SUCCESS) ? 0 : -1;
+}
+
+/*! \brief Function called by core when we should answer a 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(ast_channel_nativeformats(ast), &f->subclass.format))) {
+		ast_debug(1, "Oooh, format changed to %s\n", ast_getformatname(&f->subclass.format));
+		ast_format_cap_set(ast_channel_nativeformats(ast), &f->subclass.format);
+		ast_set_read_format(ast, ast_channel_readformat(ast));
+		ast_set_write_format(ast, ast_channel_writeformat(ast));
+	}
+
+	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(ast_channel_nativeformats(ast), &frame->subclass.format))) {
+			char buf[256];
+
+			ast_log(LOG_WARNING,
+				"Asked to transmit frame type %s, while native formats is %s (read/write = %s/%s)\n",
+				ast_getformatname(&frame->subclass.format),
+				ast_getformatname_multiple(buf, sizeof(buf), ast_channel_nativeformats(ast)),
+				ast_getformatname(ast_channel_readformat(ast)),
+				ast_getformatname(ast_channel_writeformat(ast)));
+			return 0;
+		}
+		if (media->rtp) {
+			res = ast_rtp_instance_write(media->rtp, frame);
+		}
+		break;
+	case AST_FRAME_VIDEO:
+		if ((media = pvt->media[SIP_MEDIA_VIDEO]) && media->rtp) {
+			res = ast_rtp_instance_write(media->rtp, frame);
+		}
+		break;
+	case AST_FRAME_MODEM:
+		break;
+	default:
+		ast_log(LOG_WARNING, "Can't send %d 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;
+	if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
+		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, ast_channel_uniqueid(fix_data->chan));
+	}
+	if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
+		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, 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;
+}
+
+/*! \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_caching_topic *, caching_topic, 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), 1);
+
+	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 || !(caching_topic = ast_channel_topic_all_cached())) {
+		return state;
+	}
+
+	ast_devstate_aggregate_init(&aggregate);
+
+	ao2_ref(caching_topic, +1);
+
+	for (num = 0; num < endpoint_snapshot->num_channels; num++) {
+		RAII_VAR(struct stasis_message *, msg, stasis_cache_get_extended(caching_topic, ast_channel_snapshot_type(),
+			endpoint_snapshot->channel_ids[num], 1), ao2_cleanup);
+		struct ast_channel_snapshot *snapshot;
+
+		if (!msg) {
+			continue;
+		}
+
+		snapshot = stasis_message_data(msg);
+
+		if (snapshot->state == AST_STATE_DOWN) {
+			ast_devstate_aggregate_add(&aggregate, AST_DEVICE_NOT_INUSE);
+		} else if (snapshot->state == AST_STATE_RINGING) {
+			ast_devstate_aggregate_add(&aggregate, AST_DEVICE_RINGING);
+		} else if ((snapshot->state == AST_STATE_UP) || (snapshot->state == AST_STATE_RING) ||
+			(snapshot->state == AST_STATE_BUSY)) {
+			ast_devstate_aggregate_add(&aggregate, AST_DEVICE_INUSE);
+			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;
+}
+
+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, &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;
+
+		if (session->inv_session->invite_tsx && (session->inv_session->options & PJSIP_INV_SUPPORT_UPDATE)) {
+			method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
+		}
+
+		ast_sip_session_refresh(session, NULL, NULL, NULL, method, 0);
+	}
+
+	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;
+
+	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) {
+			ao2_ref(channel->session, +1);
+
+			if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
+				ao2_cleanup(channel->session);
+			}
+		} else {
+			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:
+		ast_moh_start(ast, data, NULL);
+		break;
+	case AST_CONTROL_UNHOLD:
+		ast_moh_stop(ast);
+		break;
+	case AST_CONTROL_SRCUPDATE:
+		break;
+	case AST_CONTROL_SRCCHANGE:
+		break;
+	case 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, &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 int call(void *data)
+{
+	struct ast_sip_session *session = data;
+	pjsip_tx_data *tdata;
+
+	int res = ast_sip_session_create_invite(session, &tdata);
+
+	if (res) {
+		ast_queue_hangup(session->channel);
+	} else {
+		ast_sip_session_send_request(session, tdata);
+	}
+	ao2_ref(session, -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->session, +1);
+	if (ast_sip_push_task(channel->session->serializer, call, channel->session)) {
+		ast_log(LOG_WARNING, "Error attempting to place outbound call to call '%s'\n", dest);
+		ao2_cleanup(channel->session);
+		return -1;
+	}
+
+	return 0;
+}
+
+/*! \brief Internal function which translates from Asterisk cause codes to SIP response codes */
+static int hangup_cause2sip(int cause)
+{
+	switch (cause) {
+	case AST_CAUSE_UNALLOCATED:             /* 1 */
+	case AST_CAUSE_NO_ROUTE_DESTINATION:    /* 3 IAX2: Can't find extension in context */
+	case AST_CAUSE_NO_ROUTE_TRANSIT_NET:    /* 2 */
+		return 404;
+	case AST_CAUSE_CONGESTION:              /* 34 */
+	case AST_CAUSE_SWITCH_CONGESTION:       /* 42 */
+		return 503;
+	case AST_CAUSE_NO_USER_RESPONSE:        /* 18 */
+		return 408;
+	case AST_CAUSE_NO_ANSWER:               /* 19 */
+	case AST_CAUSE_UNREGISTERED:        /* 20 */
+		return 480;
+	case AST_CAUSE_CALL_REJECTED:           /* 21 */
+		return 403;
+	case AST_CAUSE_NUMBER_CHANGED:          /* 22 */
+		return 410;
+	case AST_CAUSE_NORMAL_UNSPECIFIED:      /* 31 */
+		return 480;
+	case AST_CAUSE_INVALID_NUMBER_FORMAT:
+		return 484;
+	case AST_CAUSE_USER_BUSY:
+		return 486;
+	case AST_CAUSE_FAILURE:
+		return 500;
+	case AST_CAUSE_FACILITY_REJECTED:       /* 29 */
+		return 501;
+	case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
+		return 503;
+	case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
+		return 502;
+	case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL:       /* Can't find codec to connect to host */
+		return 488;
+	case AST_CAUSE_INTERWORKING:    /* Unspecified Interworking issues */
+		return 500;
+	case AST_CAUSE_NOTDEFINED:
+	default:
+		ast_debug(1, "AST hangup cause %d (no match found in PJSIP)\n", cause);
+		return 0;
+	}
+
+	/* Never reached */
+	return 0;
+}
+
+struct hangup_data {
+	int cause;
+	struct ast_channel *chan;
+};
+
+static void hangup_data_destroy(void *obj)
+{
+	struct hangup_data *h_data = obj;
+
+	h_data->chan = ast_channel_unref(h_data->chan);
+}
+
+static struct hangup_data *hangup_data_alloc(int cause, struct ast_channel *chan)
+{
+	struct hangup_data *h_data = ao2_alloc(sizeof(*h_data), hangup_data_destroy);
+
+	if (!h_data) {
+		return NULL;
+	}
+
+	h_data->cause = cause;
+	h_data->chan = ast_channel_ref(chan);
+
+	return h_data;
+}
+
+/*! \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;
+	if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
+		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_AUDIO]->rtp, "");
+	}
+	if (pvt->media[SIP_MEDIA_VIDEO] && pvt->media[SIP_MEDIA_VIDEO]->rtp) {
+		ast_rtp_instance_set_channel_id(pvt->media[SIP_MEDIA_VIDEO]->rtp, "");
+	}
+	ast_channel_tech_pvt_set(ast, NULL);
+}
+
+static int hangup(void *data)
+{
+	pj_status_t status;
+	pjsip_tx_data *packet = NULL;
+	struct hangup_data *h_data = data;
+	struct ast_channel *ast = h_data->chan;
+	struct 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 &&
+		((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, 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_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, requestor ? ast_channel_linkedid(requestor) : NULL, 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_sip_create_request("MESSAGE", data->session->inv_session->dlg, data->session->endpoint, NULL, &tdata);
+	ast_sip_add_body(tdata, &body);
+	ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint);
+
+	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;
+	}
+
+	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)
+{
+	pjsip_tx_data *packet = NULL;
+
+	if (session->channel) {
+		return 0;
+	}
+
+	if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
+		if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
+			ast_sip_session_send_response(session, packet);
+		}
+
+		ast_log(LOG_ERROR, "Failed to allocate new 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 pbx_start_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
+{
+	int res;
+
+	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_SESSION_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;
+
+	if (!session->channel) {
+		return;
+	}
+
+	switch (status.code) {
+	case 180:
+		ast_queue_control(session->channel, AST_CONTROL_RINGING);
+		if (ast_channel_state(session->channel) != AST_STATE_UP) {
+			ast_setstate(session->channel, AST_STATE_RINGING);
+		}
+		break;
+	case 183:
+		ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
+		break;
+	case 200:
+		ast_queue_control(session->channel, AST_CONTROL_ANSWER);
+		break;
+	default:
+		break;
+	}
+}
+
+static int 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) {
+			ast_queue_control(session->channel, AST_CONTROL_SRCCHANGE);
+		}
+	}
+	return 0;
+}
+
+/*!
+ * \brief Load the module
+ *
+ * Module loading including tests for configuration or dependencies.
+ * This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE,
+ * or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails
+ * tests return AST_MODULE_LOAD_FAILURE. If the module can not load the
+ * configuration file or other non-critical problem return
+ * AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.
+ */
+static int load_module(void)
+{
+	if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc())) {
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	ast_format_cap_add_all_by_type(chan_pjsip_tech.capabilities, AST_FORMAT_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");
+	}
+
+	if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
+		ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
+		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);
+		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);
+		goto end;
+	}
+
+	return 0;
+
+end:
+	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)
+{
+	ast_custom_function_unregister(&media_offer_function);
+
+	ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
+	ast_sip_session_unregister_supplement(&pbx_start_supplement);
+
+	ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
+	ast_channel_unregister(&chan_pjsip_tech);
+	ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
+
+	return 0;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
+		.load = load_module,
+		.unload = unload_module,
+		.reload = reload,
+		.load_pri = AST_MODPRI_CHANNEL_DRIVER,
+	       );