func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip

This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan
to use the CHANNEL function on a chan_pjsip channel to obtain run-time
information about the channel from the PJSIP channel driver and the PJSIP
stack. This includes:
 * RTP information, including source/destination media addresses, whether or
   not the media is secure, held, and other properties.
 * RTCP information. This includes sets of parseable information, as well as
   individual statistic attriutes.
 * PJSIP information. This includes URIs, local/remote signalling addresses,
   whether or not the signalling is secure, and other properties.
 * The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT
   function to obtain more detailed endpoint information.

Review: https://reviewboard.asterisk.org/r/3038/
........

Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3

3 files changed, 1027 insertions, 0 deletions

diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c
new file mode 100644
index 000000000..ac98be526
--- /dev/null
+++ b/channels/pjsip/dialplan_functions.c
@@ -0,0 +1,893 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * 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 \verbatim Joshua Colp <jcolp@digium.com> \endverbatim
+ * \author \verbatim Matt Jordan <mjordan@digium.com> \endverbatim
+ *
+ * \ingroup functions
+ *
+ * \brief PJSIP channel dialplan functions
+ */
+
+/*** MODULEINFO
+	<support_level>core</support_level>
+ ***/
+
+/*** 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>
+<info name="PJSIPCHANNEL" language="en_US" tech="PJSIP">
+	<enumlist>
+		<enum name="rtp">
+			<para>R/O Retrieve media related information.</para>
+			<parameter name="type" required="true">
+				<para>When <replaceable>rtp</replaceable> is specified, the
+				<literal>type</literal> parameter must be provided. It specifies
+				which RTP parameter to read.</para>
+				<enumlist>
+					<enum name="src">
+						<para>Retrieve the local address for RTP.</para>
+					</enum>
+					<enum name="dest">
+						<para>Retrieve the remote address for RTP.</para>
+					</enum>
+					<enum name="direct">
+						<para>If direct media is enabled, this address is the remote address
+						used for RTP.</para>
+					</enum>
+					<enum name="secure">
+						<para>Whether or not the media stream is encrypted.</para>
+						<enumlist>
+							<enum name="0">
+								<para>The media stream is not encrypted.</para>
+							</enum>
+							<enum name="1">
+								<para>The media stream is encrypted.</para>
+							</enum>
+						</enumlist>
+					</enum>
+					<enum name="hold">
+						<para>Whether or not the media stream is currently restricted
+						due to a call hold.</para>
+						<enumlist>
+							<enum name="0">
+								<para>The media stream is not held.</para>
+							</enum>
+							<enum name="1">
+								<para>The media stream is held.</para>
+							</enum>
+						</enumlist>
+					</enum>
+				</enumlist>
+			</parameter>
+			<parameter name="media_type" required="false">
+				<para>When <replaceable>rtp</replaceable> is specified, the
+				<literal>media_type</literal> parameter may be provided. It specifies
+				which media stream the chosen RTP parameter should be retrieved
+				from.</para>
+				<enumlist>
+					<enum name="audio">
+						<para>Retrieve information from the audio media stream.</para>
+						<note><para>If not specified, <literal>audio</literal> is used
+						by default.</para></note>
+					</enum>
+					<enum name="video">
+						<para>Retrieve information from the video media stream.</para>
+					</enum>
+				</enumlist>
+			</parameter>
+		</enum>
+		<enum name="rtcp">
+			<para>R/O Retrieve RTCP statistics.</para>
+			<parameter name="statistic" required="true">
+				<para>When <replaceable>rtcp</replaceable> is specified, the
+				<literal>statistic</literal> parameter must be provided. It specifies
+				which RTCP statistic parameter to read.</para>
+				<enumlist>
+					<enum name="all">
+						<para>Retrieve a summary of all RTCP statistics.</para>
+						<para>The following data items are returned in a semi-colon
+						delineated list:</para>
+						<enumlist>
+							<enum name="ssrc">
+								<para>Our Synchronization Source identifier</para>
+							</enum>
+							<enum name="themssrc">
+								<para>Their Synchronization Source identifier</para>
+							</enum>
+							<enum name="lp">
+								<para>Our lost packet count</para>
+							</enum>
+							<enum name="rxjitter">
+								<para>Received packet jitter</para>
+							</enum>
+							<enum name="rxcount">
+								<para>Received packet count</para>
+							</enum>
+							<enum name="txjitter">
+								<para>Transmitted packet jitter</para>
+							</enum>
+							<enum name="txcount">
+								<para>Transmitted packet count</para>
+							</enum>
+							<enum name="rlp">
+								<para>Remote lost packet count</para>
+							</enum>
+							<enum name="rtt">
+								<para>Round trip time</para>
+							</enum>
+						</enumlist>
+					</enum>
+					<enum name="all_jitter">
+						<para>Retrieve a summary of all RTCP Jitter statistics.</para>
+						<para>The following data items are returned in a semi-colon
+						delineated list:</para>
+						<enumlist>
+							<enum name="minrxjitter">
+								<para>Our minimum jitter</para>
+							</enum>
+							<enum name="maxrxjitter">
+								<para>Our max jitter</para>
+							</enum>
+							<enum name="avgrxjitter">
+								<para>Our average jitter</para>
+							</enum>
+							<enum name="stdevrxjitter">
+								<para>Our jitter standard deviation</para>
+							</enum>
+							<enum name="reported_minjitter">
+								<para>Their minimum jitter</para>
+							</enum>
+							<enum name="reported_maxjitter">
+								<para>Their max jitter</para>
+							</enum>
+							<enum name="reported_avgjitter">
+								<para>Their average jitter</para>
+							</enum>
+							<enum name="reported_stdevjitter">
+								<para>Their jitter standard deviation</para>
+							</enum>
+						</enumlist>
+					</enum>
+					<enum name="all_loss">
+						<para>Retrieve a summary of all RTCP packet loss statistics.</para>
+						<para>The following data items are returned in a semi-colon
+						delineated list:</para>
+						<enumlist>
+							<enum name="minrxlost">
+								<para>Our minimum lost packets</para>
+							</enum>
+							<enum name="maxrxlost">
+								<para>Our max lost packets</para>
+							</enum>
+							<enum name="avgrxlost">
+								<para>Our average lost packets</para>
+							</enum>
+							<enum name="stdevrxlost">
+								<para>Our lost packets standard deviation</para>
+							</enum>
+							<enum name="reported_minlost">
+								<para>Their minimum lost packets</para>
+							</enum>
+							<enum name="reported_maxlost">
+								<para>Their max lost packets</para>
+							</enum>
+							<enum name="reported_avglost">
+								<para>Their average lost packets</para>
+							</enum>
+							<enum name="reported_stdevlost">
+								<para>Their lost packets standard deviation</para>
+							</enum>
+						</enumlist>
+					</enum>
+					<enum name="all_rtt">
+						<para>Retrieve a summary of all RTCP round trip time information.</para>
+						<para>The following data items are returned in a semi-colon
+						delineated list:</para>
+						<enumlist>
+							<enum name="minrtt">
+								<para>Minimum round trip time</para>
+							</enum>
+							<enum name="maxrtt">
+								<para>Maximum round trip time</para>
+							</enum>
+							<enum name="avgrtt">
+								<para>Average round trip time</para>
+							</enum>
+							<enum name="stdevrtt">
+								<para>Standard deviation round trip time</para>
+							</enum>
+						</enumlist>
+					</enum>
+					<enum name="txcount"><para>Transmitted packet count</para></enum>
+					<enum name="rxcount"><para>Received packet count</para></enum>
+					<enum name="txjitter"><para>Transmitted packet jitter</para></enum>
+					<enum name="rxjitter"><para>Received packet jitter</para></enum>
+					<enum name="remote_maxjitter"><para>Their max jitter</para></enum>
+					<enum name="remote_minjitter"><para>Their minimum jitter</para></enum>
+					<enum name="remote_normdevjitter"><para>Their average jitter</para></enum>
+					<enum name="remote_stdevjitter"><para>Their jitter standard deviation</para></enum>
+					<enum name="local_maxjitter"><para>Our max jitter</para></enum>
+					<enum name="local_minjitter"><para>Our minimum jitter</para></enum>
+					<enum name="local_normdevjitter"><para>Our average jitter</para></enum>
+					<enum name="local_stdevjitter"><para>Our jitter standard deviation</para></enum>
+					<enum name="txploss"><para>Transmitted packet loss</para></enum>
+					<enum name="rxploss"><para>Received packet loss</para></enum>
+					<enum name="remote_maxrxploss"><para>Their max lost packets</para></enum>
+					<enum name="remote_minrxploss"><para>Their minimum lost packets</para></enum>
+					<enum name="remote_normdevrxploss"><para>Their average lost packets</para></enum>
+					<enum name="remote_stdevrxploss"><para>Their lost packets standard deviation</para></enum>
+					<enum name="local_maxrxploss"><para>Our max lost packets</para></enum>
+					<enum name="local_minrxploss"><para>Our minimum lost packets</para></enum>
+					<enum name="local_normdevrxploss"><para>Our average lost packets</para></enum>
+					<enum name="local_stdevrxploss"><para>Our lost packets standard deviation</para></enum>
+					<enum name="rtt"><para>Round trip time</para></enum>
+					<enum name="maxrtt"><para>Maximum round trip time</para></enum>
+					<enum name="minrtt"><para>Minimum round trip time</para></enum>
+					<enum name="normdevrtt"><para>Average round trip time</para></enum>
+					<enum name="stdevrtt"><para>Standard deviation round trip time</para></enum>
+					<enum name="local_ssrc"><para>Our Synchronization Source identifier</para></enum>
+					<enum name="remote_ssrc"><para>Their Synchronization Source identifier</para></enum>
+				</enumlist>
+			</parameter>
+			<parameter name="media_type" required="false">
+				<para>When <replaceable>rtcp</replaceable> is specified, the
+				<literal>media_type</literal> parameter may be provided. It specifies
+				which media stream the chosen RTCP parameter should be retrieved
+				from.</para>
+				<enumlist>
+					<enum name="audio">
+						<para>Retrieve information from the audio media stream.</para>
+						<note><para>If not specified, <literal>audio</literal> is used
+						by default.</para></note>
+					</enum>
+					<enum name="video">
+						<para>Retrieve information from the video media stream.</para>
+					</enum>
+				</enumlist>
+			</parameter>
+		</enum>
+		<enum name="endpoint">
+			<para>R/O The name of the endpoint associated with this channel.
+			Use the <replaceable>PJSIP_ENDPOINT</replaceable> function to obtain
+			further endpoint related information.</para>
+		</enum>
+		<enum name="pjsip">
+			<para>R/O Obtain information about the current PJSIP channel and its
+			session.</para>
+			<parameter name="type" required="true">
+				<para>When <replaceable>pjsip</replaceable> is specified, the
+				<literal>type</literal> parameter must be provided. It specifies
+				which signalling parameter to read.</para>
+				<enumlist>
+					<enum name="secure">
+						<para>Whether or not the signalling uses a secure transport.</para>
+						<enumlist>
+							<enum name="0"><para>The signalling uses a non-secure transport.</para></enum>
+							<enum name="1"><para>The signalling uses a secure transport.</para></enum>
+						</enumlist>
+					</enum>
+					<enum name="target_uri">
+						<para>The request URI of the <literal>INVITE</literal> request associated with the creation of this channel.</para>
+					</enum>
+					<enum name="local_uri">
+						<para>The local URI.</para>
+					</enum>
+					<enum name="remote_uri">
+						<para>The remote URI.</para>
+					</enum>
+					<enum name="t38state">
+						<para>The current state of any T.38 fax on this channel.</para>
+						<enumlist>
+							<enum name="DISABLED"><para>T.38 faxing is disabled on this channel.</para></enum>
+							<enum name="LOCAL_REINVITE"><para>Asterisk has sent a <literal>re-INVITE</literal> to the remote end to initiate a T.38 fax.</para></enum>
+							<enum name="REMOTE_REINVITE"><para>The remote end has sent a <literal>re-INVITE</literal> to Asterisk to initiate a T.38 fax.</para></enum>
+							<enum name="ENABLED"><para>A T.38 fax session has been enabled.</para></enum>
+							<enum name="REJECTED"><para>A T.38 fax session was attempted but was rejected.</para></enum>
+						</enumlist>
+					</enum>
+					<enum name="local_addr">
+						<para>On inbound calls, the full IP address and port number that
+						the <literal>INVITE</literal> request was received on. On outbound
+						calls, the full IP address and port number that the <literal>INVITE</literal>
+						request was transmitted from.</para>
+					</enum>
+					<enum name="remote_addr">
+						<para>On inbound calls, the full IP address and port number that
+						the <literal>INVITE</literal> request was received from. On outbound
+						calls, the full IP address and port number that the <literal>INVITE</literal>
+						request was transmitted to.</para>
+					</enum>
+				</enumlist>
+			</parameter>
+		</enum>
+	</enumlist>
+</info>
+***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+#include <pjlib.h>
+#include <pjsip_ua.h>
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include "asterisk/astobj2.h"
+#include "asterisk/module.h"
+#include "asterisk/acl.h"
+#include "asterisk/app.h"
+#include "asterisk/channel.h"
+#include "asterisk/format.h"
+#include "asterisk/pbx.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
+#include "include/chan_pjsip.h"
+#include "include/dialplan_functions.h"
+
+/*!
+ * \brief String representations of the T.38 state enum
+ */
+static const char *t38state_to_string[T38_MAX_ENUM] = {
+	[T38_DISABLED] = "DISABLED",
+	[T38_LOCAL_REINVITE] = "LOCAL_REINVITE",
+	[T38_PEER_REINVITE] = "REMOTE_REINVITE",
+	[T38_ENABLED] = "ENABLED",
+	[T38_REJECTED] = "REJECTED",
+};
+
+/*!
+ * \internal \brief Handle reading RTP information
+ */
+static int channel_read_rtp(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
+{
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	struct chan_pjsip_pvt *pvt;
+	struct ast_sip_session_media *media = NULL;
+	struct ast_sockaddr addr;
+
+	if (!channel) {
+		ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
+		return -1;
+	}
+
+	pvt = channel->pvt;
+	if (!pvt) {
+		ast_log(AST_LOG_WARNING, "Channel %s has no chan_pjsip pvt!\n", ast_channel_name(chan));
+		return -1;
+	}
+
+	if (ast_strlen_zero(type)) {
+		ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtp' information\n");
+		return -1;
+	}
+
+	if (ast_strlen_zero(field) || !strcmp(field, "audio")) {
+		media = pvt->media[SIP_MEDIA_AUDIO];
+	} else if (!strcmp(field, "video")) {
+		media = pvt->media[SIP_MEDIA_VIDEO];
+	} else {
+		ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtp' information\n", field);
+		return -1;
+	}
+
+	if (!media || !media->rtp) {
+		ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
+			ast_channel_name(chan), S_OR(field, "audio"));
+		return -1;
+	}
+
+	if (!strcmp(type, "src")) {
+		ast_rtp_instance_get_local_address(media->rtp, &addr);
+		ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
+	} else if (!strcmp(type, "dest")) {
+		ast_rtp_instance_get_remote_address(media->rtp, &addr);
+		ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
+	} else if (!strcmp(type, "direct")) {
+		ast_copy_string(buf, ast_sockaddr_stringify(&media->direct_media_addr), buflen);
+	} else if (!strcmp(type, "secure")) {
+		snprintf(buf, buflen, "%u", media->srtp ? 1 : 0);
+	} else if (!strcmp(type, "hold")) {
+		snprintf(buf, buflen, "%u", media->held ? 1 : 0);
+	} else {
+		ast_log(AST_LOG_WARNING, "Unknown type field '%s' specified for 'rtp' information\n", type);
+		return -1;
+	}
+
+	return 0;
+}
+
+/*!
+ * \internal \brief Handle reading RTCP information
+ */
+static int channel_read_rtcp(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
+{
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	struct chan_pjsip_pvt *pvt;
+	struct ast_sip_session_media *media = NULL;
+
+	if (!channel) {
+		ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
+		return -1;
+	}
+
+	pvt = channel->pvt;
+	if (!pvt) {
+		ast_log(AST_LOG_WARNING, "Channel %s has no chan_pjsip pvt!\n", ast_channel_name(chan));
+		return -1;
+	}
+
+	if (ast_strlen_zero(type)) {
+		ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtcp' information\n");
+		return -1;
+	}
+
+	if (ast_strlen_zero(field) || !strcmp(field, "audio")) {
+		media = pvt->media[SIP_MEDIA_AUDIO];
+	} else if (!strcmp(field, "video")) {
+		media = pvt->media[SIP_MEDIA_VIDEO];
+	} else {
+		ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtcp' information\n", field);
+		return -1;
+	}
+
+	if (!media || !media->rtp) {
+		ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
+			ast_channel_name(chan), S_OR(field, "audio"));
+		return -1;
+	}
+
+	if (!strncasecmp(type, "all", 3)) {
+		enum ast_rtp_instance_stat_field stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY;
+
+		if (!strcasecmp(type, "all_jitter")) {
+			stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER;
+		} else if (!strcasecmp(type, "all_rtt")) {
+			stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT;
+		} else if (!strcasecmp(type, "all_loss")) {
+			stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS;
+		}
+
+		if (!ast_rtp_instance_get_quality(media->rtp, stat_field, buf, buflen)) {
+			ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
+			return -1;
+		}
+	} else {
+		struct ast_rtp_instance_stats stats;
+		int i;
+		struct {
+			const char *name;
+			enum { INT, DBL } type;
+			union {
+				unsigned int *i4;
+				double *d8;
+			};
+		} lookup[] = {
+			{ "txcount",               INT, { .i4 = &stats.txcount, }, },
+			{ "rxcount",               INT, { .i4 = &stats.rxcount, }, },
+			{ "txjitter",              DBL, { .d8 = &stats.txjitter, }, },
+			{ "rxjitter",              DBL, { .d8 = &stats.rxjitter, }, },
+			{ "remote_maxjitter",      DBL, { .d8 = &stats.remote_maxjitter, }, },
+			{ "remote_minjitter",      DBL, { .d8 = &stats.remote_minjitter, }, },
+			{ "remote_normdevjitter",  DBL, { .d8 = &stats.remote_normdevjitter, }, },
+			{ "remote_stdevjitter",    DBL, { .d8 = &stats.remote_stdevjitter, }, },
+			{ "local_maxjitter",       DBL, { .d8 = &stats.local_maxjitter, }, },
+			{ "local_minjitter",       DBL, { .d8 = &stats.local_minjitter, }, },
+			{ "local_normdevjitter",   DBL, { .d8 = &stats.local_normdevjitter, }, },
+			{ "local_stdevjitter",     DBL, { .d8 = &stats.local_stdevjitter, }, },
+			{ "txploss",               INT, { .i4 = &stats.txploss, }, },
+			{ "rxploss",               INT, { .i4 = &stats.rxploss, }, },
+			{ "remote_maxrxploss",     DBL, { .d8 = &stats.remote_maxrxploss, }, },
+			{ "remote_minrxploss",     DBL, { .d8 = &stats.remote_minrxploss, }, },
+			{ "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, },
+			{ "remote_stdevrxploss",   DBL, { .d8 = &stats.remote_stdevrxploss, }, },
+			{ "local_maxrxploss",      DBL, { .d8 = &stats.local_maxrxploss, }, },
+			{ "local_minrxploss",      DBL, { .d8 = &stats.local_minrxploss, }, },
+			{ "local_normdevrxploss",  DBL, { .d8 = &stats.local_normdevrxploss, }, },
+			{ "local_stdevrxploss",    DBL, { .d8 = &stats.local_stdevrxploss, }, },
+			{ "rtt",                   DBL, { .d8 = &stats.rtt, }, },
+			{ "maxrtt",                DBL, { .d8 = &stats.maxrtt, }, },
+			{ "minrtt",                DBL, { .d8 = &stats.minrtt, }, },
+			{ "normdevrtt",            DBL, { .d8 = &stats.normdevrtt, }, },
+			{ "stdevrtt",              DBL, { .d8 = &stats.stdevrtt, }, },
+			{ "local_ssrc",            INT, { .i4 = &stats.local_ssrc, }, },
+			{ "remote_ssrc",           INT, { .i4 = &stats.remote_ssrc, }, },
+			{ NULL, },
+		};
+
+		if (ast_rtp_instance_get_stats(media->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+			ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
+			return -1;
+		}
+
+		for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
+			if (!strcasecmp(type, lookup[i].name)) {
+				if (lookup[i].type == INT) {
+					snprintf(buf, buflen, "%u", *lookup[i].i4);
+				} else {
+					snprintf(buf, buflen, "%f", *lookup[i].d8);
+				}
+				return 0;
+			}
+		}
+		ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'rtcp' information\n", type);
+		return -1;
+	}
+
+	return 0;
+}
+
+/*!
+ * \internal \brief Handle reading signalling information
+ */
+static int channel_read_pjsip(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
+{
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	char *buf_copy;
+	pjsip_dialog *dlg;
+
+	if (!channel) {
+		ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
+		return -1;
+	}
+
+	dlg = channel->session->inv_session->dlg;
+
+	if (!strcmp(type, "secure")) {
+		snprintf(buf, buflen, "%u", dlg->secure ? 1 : 0);
+	} else if (!strcmp(type, "target_uri")) {
+		pjsip_uri_print(PJSIP_URI_IN_REQ_URI, dlg->target, buf, sizeof(buflen));
+		buf_copy = ast_strdupa(buf);
+		ast_escape_quoted(buf_copy, buf, buflen);
+	} else if (!strcmp(type, "local_uri")) {
+		pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, dlg->local.info->uri, buf, sizeof(buflen));
+		buf_copy = ast_strdupa(buf);
+		ast_escape_quoted(buf_copy, buf, buflen);
+	} else if (!strcmp(type, "remote_uri")) {
+		pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, dlg->remote.info->uri, buf, sizeof(buflen));
+		buf_copy = ast_strdupa(buf);
+		ast_escape_quoted(buf_copy, buf, buflen);
+	} else if (!strcmp(type, "t38state")) {
+		ast_copy_string(buf, t38state_to_string[channel->session->t38state], buflen);
+	} else if (!strcmp(type, "local_addr")) {
+		RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
+		struct transport_info_data *transport_data;
+
+		datastore = ast_sip_session_get_datastore(channel->session, "transport_info");
+		if (!datastore) {
+			ast_log(AST_LOG_WARNING, "No transport information for channel %s\n", ast_channel_name(chan));
+			return -1;
+		}
+		transport_data = datastore->data;
+
+		if (pj_sockaddr_has_addr(&transport_data->local_addr)) {
+			pj_sockaddr_print(&transport_data->local_addr, buf, buflen, 3);
+		}
+	} else if (!strcmp(type, "remote_addr")) {
+		RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
+		struct transport_info_data *transport_data;
+
+		datastore = ast_sip_session_get_datastore(channel->session, "transport_info");
+		if (!datastore) {
+			ast_log(AST_LOG_WARNING, "No transport information for channel %s\n", ast_channel_name(chan));
+			return -1;
+		}
+		transport_data = datastore->data;
+
+		if (pj_sockaddr_has_addr(&transport_data->remote_addr)) {
+			pj_sockaddr_print(&transport_data->remote_addr, buf, buflen, 3);
+		}
+	} else {
+		ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'pjsip' information\n", type);
+		return -1;
+	}
+
+	return 0;
+}
+
+/*! \brief Struct used to push function arguments to task processor */
+struct pjsip_func_args {
+	struct ast_channel *chan;
+	const char *param;
+	const char *type;
+	const char *field;
+	char *buf;
+	size_t len;
+	int ret;
+};
+
+/*! \internal \brief Taskprocessor callback that handles the read on a PJSIP thread */
+static int read_pjsip(void *data)
+{
+	struct pjsip_func_args *func_args = data;
+
+	if (!strcmp(func_args->param, "rtp")) {
+		func_args->ret = channel_read_rtp(func_args->chan, func_args->type,
+		                                  func_args->field, func_args->buf,
+		                                  func_args->len);
+	} else if (!strcmp(func_args->param, "rtcp")) {
+		func_args->ret = channel_read_rtcp(func_args->chan, func_args->type,
+		                                   func_args->field, func_args->buf,
+		                                   func_args->len);
+	} else if (!strcmp(func_args->param, "endpoint")) {
+		struct ast_sip_channel_pvt *pvt = ast_channel_tech_pvt(func_args->chan);
+
+		if (!pvt) {
+			ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(func_args->chan));
+			return -1;
+		}
+		if (!pvt->session || !pvt->session->endpoint) {
+			ast_log(AST_LOG_WARNING, "Channel %s has no endpoint!\n", ast_channel_name(func_args->chan));
+			return -1;
+		}
+		snprintf(func_args->buf, func_args->len, "%s", ast_sorcery_object_get_id(pvt->session->endpoint));
+	} else if (!strcmp(func_args->param, "pjsip")) {
+		func_args->ret = channel_read_pjsip(func_args->chan, func_args->type,
+		                                    func_args->field, func_args->buf,
+		                                    func_args->len);
+	} else {
+		func_args->ret = -1;
+	}
+
+	return 0;
+}
+
+
+int pjsip_acf_channel_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+	struct pjsip_func_args func_args = { 0, };
+	struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
+	char *parse = ast_strdupa(data);
+
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(param);
+		AST_APP_ARG(type);
+		AST_APP_ARG(field);
+	);
+
+	/* Check for zero arguments */
+	if (ast_strlen_zero(parse)) {
+		ast_log(LOG_ERROR, "Cannot call %s without arguments\n", cmd);
+		return -1;
+	}
+
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	/* Sanity check */
+	if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+		ast_log(LOG_ERROR, "Cannot call %s on a non-PJSIP channel\n", cmd);
+		return 0;
+	}
+
+	if (!channel) {
+		ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
+		return -1;
+	}
+
+	memset(buf, 0, len);
+
+	func_args.chan = chan;
+	func_args.param = args.param;
+	func_args.type = args.type;
+	func_args.field = args.field;
+	func_args.buf = buf;
+	func_args.len = len;
+	if (ast_sip_push_task_synchronous(channel->session->serializer, read_pjsip, &func_args)) {
+		ast_log(LOG_WARNING, "Unable to read properties of channel %s: failed to push task\n", ast_channel_name(chan));
+		return -1;
+	}
+
+	return func_args.ret;
+}
+
+int pjsip_acf_dial_contacts_read(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 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;
+}
+
+int pjsip_acf_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;
+}
+
+int pjsip_acf_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);
+}
diff --git a/channels/pjsip/include/chan_pjsip.h b/channels/pjsip/include/chan_pjsip.h
new file mode 100644
index 000000000..b229a0487
--- /dev/null
+++ b/channels/pjsip/include/chan_pjsip.h
@@ -0,0 +1,58 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * 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
+ * \brief PJSIP Channel Driver shared data structures
+ */
+
+#ifndef _CHAN_PJSIP_HEADER
+#define _CHAN_PJSIP_HEADER
+
+struct ast_sip_session_media;
+
+/*!
+ * \brief Transport information stored in transport_info datastore
+ */
+struct transport_info_data {
+	/*! \brief The address that sent the request */
+	pj_sockaddr remote_addr;
+	/*! \brief Our address that received the request */
+	pj_sockaddr local_addr;
+};
+
+/*!
+ * \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,
+};
+
+/*!
+ * \brief The PJSIP channel driver pvt, stored in the \ref ast_sip_channel_pvt
+ * data structure
+ */
+struct chan_pjsip_pvt {
+	/*! \brief The available media sessions */
+	struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
+};
+
+#endif /* _CHAN_PJSIP_HEADER */
diff --git a/channels/pjsip/include/dialplan_functions.h b/channels/pjsip/include/dialplan_functions.h
new file mode 100644
index 000000000..cbc06f076
--- /dev/null
+++ b/channels/pjsip/include/dialplan_functions.h
@@ -0,0 +1,76 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * 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
+ * \brief PJSIP dialplan functions header file
+ */
+
+#ifndef _PJSIP_DIALPLAN_FUNCTIONS
+#define _PJSIP_DIALPLAN_FUNCTIONS
+
+/*!
+ * \brief CHANNEL function read callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param buf Out buffer that should be populated with the data
+ * \param len Size of the buffer
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_channel_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
+
+/*!
+ * \brief PJSIP_MEDIA_OFFER function write callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param value Value to be set by the function
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
+
+/*!
+ * \brief PJSIP_MEDIA_OFFER function read callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param buf Out buffer that should be populated with the data
+ * \param len Size of the buffer
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
+
+/*!
+ * \brief PJSIP_DIAL_CONTACTS function read callback
+ * \param chan The channel the function is called on
+ * \param cmd The name of the function
+ * \param data Arguments passed to the function
+ * \param buf Out buffer that should be populated with the data
+ * \param len Size of the buffer
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_acf_dial_contacts_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
+
+#endif /* _PJSIP_DIALPLAN_FUNCTIONS */
\ No newline at end of file