func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip
authorMatthew Jordan <mjordan@digium.com>
Wed, 11 Dec 2013 13:06:30 +0000 (13:06 +0000)
committerMatthew Jordan <mjordan@digium.com>
Wed, 11 Dec 2013 13:06:30 +0000 (13:06 +0000)
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

channels/Makefile
channels/chan_pjsip.c
channels/pjsip/dialplan_functions.c [new file with mode: 0644]
channels/pjsip/include/chan_pjsip.h [new file with mode: 0644]
channels/pjsip/include/dialplan_functions.h [new file with mode: 0644]
funcs/func_channel.c
include/asterisk/res_pjsip_session.h
main/xmldoc.c
res/res_pjsip_t38.c

index a30daa5..79404d6 100644 (file)
@@ -77,6 +77,9 @@ $(subst .c,.o,$(wildcard iax2/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,chan_iax2)
 $(if $(filter chan_sip,$(EMBEDDED_MODS)),modules.link,chan_sip.so): $(subst .c,.o,$(wildcard sip/*.c))
 $(subst .c,.o,$(wildcard sip/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,chan_sip)
 
+$(if $(filter chan_pjsip,$(EMBEDDED_MODS)),modules.link,chan_pjsip.so): $(subst .c,.o,$(wildcard pjsip/*.c))
+$(subst .c,.o,$(wildcard pjsip/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,chan_pjsip)
+
 # Additional objects to combine with chan_dahdi.so
 CHAN_DAHDI_OBJS= \
        $(subst .c,.o,$(wildcard dahdi/*.c))    \
index ad74b57..71edb35 100644 (file)
@@ -61,62 +61,14 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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>
- ***/
+#include "pjsip/include/chan_pjsip.h"
+#include "pjsip/include/dialplan_functions.h"
 
 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;
@@ -145,7 +97,7 @@ 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 = {
+struct ast_channel_tech chan_pjsip_tech = {
        .type = channel_type,
        .description = "PJSIP Channel Driver",
        .requester = chan_pjsip_request,
@@ -164,6 +116,7 @@ static struct ast_channel_tech chan_pjsip_tech = {
        .fixup = chan_pjsip_fixup,
        .devicestate = chan_pjsip_devicestate,
        .queryoption = chan_pjsip_queryoption,
+       .func_channel_read = pjsip_acf_channel_read,
        .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
 };
 
@@ -191,184 +144,6 @@ static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
        .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)
 {
@@ -437,6 +212,20 @@ static int send_direct_media_request(void *data)
                        session->endpoint->media.direct_media.method, 1);
 }
 
+/*! \brief Destructor function for \ref transport_info_data */
+static void transport_info_destroy(void *obj)
+{
+       struct transport_info_data *data = obj;
+       ast_free(data);
+}
+
+/*! \brief Datastore used to store local/remote addresses for the
+ * INVITE request that created the PJSIP channel */
+static struct ast_datastore_info transport_info = {
+       .type = "chan_pjsip_transport_info",
+       .destroy = transport_info_destroy,
+};
+
 static struct ast_datastore_info direct_media_mitigation_info = { };
 
 static int direct_media_mitigate_glare(struct ast_sip_session *session)
@@ -1989,12 +1778,28 @@ static void chan_pjsip_session_end(struct ast_sip_session *session)
 /*! \brief Function called when a request is received on the session */
 static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
 {
+       RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
+       struct transport_info_data *transport_data;
        pjsip_tx_data *packet = NULL;
 
        if (session->channel) {
                return 0;
        }
 
+       datastore = ast_sip_session_alloc_datastore(&transport_info, "transport_info");
+       if (!datastore) {
+               return -1;
+       }
+
+       transport_data = ast_calloc(1, sizeof(*transport_data));
+       if (!transport_data) {
+               return -1;
+       }
+       pj_sockaddr_cp(&transport_data->local_addr, &rdata->tp_info.transport->local_addr);
+       pj_sockaddr_cp(&transport_data->remote_addr, &rdata->pkt_info.src_addr);
+       datastore->data = transport_data;
+       ast_sip_session_add_datastore(session, datastore);
+
        if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
                if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
                        ast_sip_session_send_response(session, packet);
@@ -2078,6 +1883,17 @@ static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip
        return 0;
 }
 
+static struct ast_custom_function chan_pjsip_dial_contacts_function = {
+       .name = "PJSIP_DIAL_CONTACTS",
+       .read = pjsip_acf_dial_contacts_read,
+};
+
+static struct ast_custom_function media_offer_function = {
+       .name = "PJSIP_MEDIA_OFFER",
+       .read = pjsip_acf_media_offer_read,
+       .write = pjsip_acf_media_offer_write
+};
+
 /*!
  * \brief Load the module
  *
@@ -2110,6 +1926,7 @@ static int load_module(void)
 
        if (ast_custom_function_register(&media_offer_function)) {
                ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
+               goto end;
        }
 
        if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
@@ -2150,13 +1967,13 @@ static int reload(void)
 /*! \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_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
 
+       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);
 
diff --git a/channels/pjsip/dialplan_functions.c b/channels/pjsip/dialplan_functions.c
new file mode 100644 (file)
index 0000000..ac98be5
--- /dev/null
@@ -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 (file)
index 0000000..b229a04
--- /dev/null
@@ -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 (file)
index 0000000..cbc06f0
--- /dev/null
@@ -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
index ccdbb6e..af9a6a9 100644 (file)
@@ -280,6 +280,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
                                                <para>   Defaults to <literal>audio</literal> if unspecified.</para>
                                        </enum>
                                </enumlist>
+                               <xi:include xpointer="xpointer(/docs/info[@name='PJSIPCHANNEL'])" />
                                <para><emphasis>chan_iax2</emphasis> provides the following additional options:</para>
                                <enumlist>
                                        <enum name="osptoken">
index 7b359cd..615a621 100644 (file)
@@ -52,6 +52,7 @@ enum ast_sip_session_t38state {
        T38_PEER_REINVITE,  /*!< Offered from peer - REINVITE */
        T38_ENABLED,        /*!< Negotiated (enabled) */
        T38_REJECTED,       /*!< Refused */
+       T38_MAX_ENUM,       /*!< Not an actual state; used as max value in the enum */
 };
 
 struct ast_sip_session_sdp_handler;
index 8a54d6f..7a48c87 100644 (file)
@@ -70,6 +70,7 @@ struct documentation_tree {
 
 static char *xmldoc_get_syntax_cmd(struct ast_xml_node *fixnode, const char *name, int printname);
 static int xmldoc_parse_enumlist(struct ast_xml_node *fixnode, const char *tabs, struct ast_str **buffer);
+static void xmldoc_parse_parameter(struct ast_xml_node *fixnode, const char *tabs, struct ast_str **buffer);
 static int xmldoc_parse_info(struct ast_xml_node *node, const char *tabs, const char *posttabs, struct ast_str **buffer);
 static int xmldoc_parse_para(struct ast_xml_node *node, const char *tabs, const char *posttabs, struct ast_str **buffer);
 static int xmldoc_parse_specialtags(struct ast_xml_node *fixnode, const char *tabs, const char *posttabs, struct ast_str **buffer);
@@ -1492,58 +1493,6 @@ static int xmldoc_parse_specialtags(struct ast_xml_node *fixnode, const char *ta
 
 /*!
  * \internal
- * \brief Parse an 'info' tag inside an element.
- *
- * \param node A pointer to the 'info' xml node.
- * \param tabs A string to be appended at the beginning of each line being printed
- *             inside 'buffer'
- * \param posttabs Add this string after the content of the <para> element, if one exists
- * \param String buffer to put values found inide the info element.
- *
- * \retval 2 if the information contained a para element, and it returned a value of 2
- * \retval 1 if information was put into the buffer
- * \retval 0 if no information was put into the buffer or error
- */
-static int xmldoc_parse_info(struct ast_xml_node *node, const char *tabs, const char *posttabs, struct ast_str **buffer)
-{
-       const char *tech;
-       char *internaltabs;
-       int internal_ret;
-       int ret = 0;
-
-       if (strcasecmp(ast_xml_node_get_name(node), "info")) {
-               return ret;
-       }
-
-       ast_asprintf(&internaltabs, "%s    ", tabs);
-       if (!internaltabs) {
-               return ret;
-       }
-
-       tech = ast_xml_get_attribute(node, "tech");
-       if (tech) {
-               ast_str_append(buffer, 0, "%s<note>Technology: %s</note>\n", internaltabs, tech);
-               ast_xml_free_attr(tech);
-       }
-
-       ret = 1;
-
-       for (node = ast_xml_node_get_children(node); node; node = ast_xml_node_get_next(node)) {
-               if (!strcasecmp(ast_xml_node_get_name(node), "enumlist")) {
-                       xmldoc_parse_enumlist(node, internaltabs, buffer);
-               } else if ((internal_ret = xmldoc_parse_common_elements(node, internaltabs, posttabs, buffer))) {
-                       if (internal_ret > ret) {
-                               ret = internal_ret;
-                       }
-               }
-       }
-       ast_free(internaltabs);
-
-       return ret;
-}
-
-/*!
- * \internal
  * \brief Parse an <argument> element from the xml documentation.
  *
  * \param fixnode Pointer to the 'argument' xml node.
@@ -1829,6 +1778,7 @@ static int xmldoc_parse_enum(struct ast_xml_node *fixnode, const char *tabs, str
                }
 
                xmldoc_parse_enumlist(node, optiontabs, buffer);
+               xmldoc_parse_parameter(node, optiontabs, buffer);
        }
 
        ast_free(optiontabs);
@@ -2053,6 +2003,60 @@ static void xmldoc_parse_parameter(struct ast_xml_node *fixnode, const char *tab
 
 /*!
  * \internal
+ * \brief Parse an 'info' tag inside an element.
+ *
+ * \param node A pointer to the 'info' xml node.
+ * \param tabs A string to be appended at the beginning of each line being printed
+ *             inside 'buffer'
+ * \param posttabs Add this string after the content of the <para> element, if one exists
+ * \param String buffer to put values found inide the info element.
+ *
+ * \retval 2 if the information contained a para element, and it returned a value of 2
+ * \retval 1 if information was put into the buffer
+ * \retval 0 if no information was put into the buffer or error
+ */
+static int xmldoc_parse_info(struct ast_xml_node *node, const char *tabs, const char *posttabs, struct ast_str **buffer)
+{
+       const char *tech;
+       char *internaltabs;
+       int internal_ret;
+       int ret = 0;
+
+       if (strcasecmp(ast_xml_node_get_name(node), "info")) {
+               return ret;
+       }
+
+       ast_asprintf(&internaltabs, "%s    ", tabs);
+       if (!internaltabs) {
+               return ret;
+       }
+
+       tech = ast_xml_get_attribute(node, "tech");
+       if (tech) {
+               ast_str_append(buffer, 0, "%s<note>Technology: %s</note>\n", internaltabs, tech);
+               ast_xml_free_attr(tech);
+       }
+
+       ret = 1;
+
+       for (node = ast_xml_node_get_children(node); node; node = ast_xml_node_get_next(node)) {
+               if (!strcasecmp(ast_xml_node_get_name(node), "enumlist")) {
+                       xmldoc_parse_enumlist(node, internaltabs, buffer);
+               } else if (!strcasecmp(ast_xml_node_get_name(node), "parameter")) {
+                       xmldoc_parse_parameter(node, internaltabs, buffer);
+               } else if ((internal_ret = xmldoc_parse_common_elements(node, internaltabs, posttabs, buffer))) {
+                       if (internal_ret > ret) {
+                               ret = internal_ret;
+                       }
+               }
+       }
+       ast_free(internaltabs);
+
+       return ret;
+}
+
+/*!
+ * \internal
  * \brief Build the arguments for an item
  *
  * \param node The arguments node to parse
index 3c97784..afe1250 100644 (file)
@@ -172,6 +172,10 @@ static void t38_change_state(struct ast_sip_session *session, struct ast_sip_ses
        case T38_LOCAL_REINVITE:
                /* wait until we get a peer response before responding to local reinvite */
                break;
+       case T38_MAX_ENUM:
+               /* Well, that shouldn't happen */
+               ast_assert(0);
+               break;
        }
 
        if (parameters.request_response) {