diff --git a/channels/Makefile b/channels/Makefile
index a30daa5bad..79404d6b6b 100644
--- a/channels/Makefile
+++ b/channels/Makefile
@@ -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)) \
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index ad74b574e0..71edb351b5 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -61,62 +61,14 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
-/*** DOCUMENTATION
-
-
- Return a dial string for dialing all contacts on an AOR.
-
-
-
- Name of the endpoint
-
-
- Name of an AOR to use, if not specified the configured AORs on the endpoint are used
-
-
- Optional request user to use in the request URI
-
-
-
- Returns a properly formatted dial string for dialing all contacts on an AOR.
-
-
-
-
- Media and codec offerings to be set on an outbound SIP channel prior to dialing.
-
-
-
- types of media offered
-
-
-
- Returns the codecs offered based upon the media choice
-
-
- ***/
+#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
index 0000000000..ac98be5261
--- /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 \endverbatim
+ * \author \verbatim Matt Jordan \endverbatim
+ *
+ * \ingroup functions
+ *
+ * \brief PJSIP channel dialplan functions
+ */
+
+/*** MODULEINFO
+ core
+ ***/
+
+/*** DOCUMENTATION
+
+
+ Return a dial string for dialing all contacts on an AOR.
+
+
+
+ Name of the endpoint
+
+
+ Name of an AOR to use, if not specified the configured AORs on the endpoint are used
+
+
+ Optional request user to use in the request URI
+
+
+
+ Returns a properly formatted dial string for dialing all contacts on an AOR.
+
+
+
+
+ Media and codec offerings to be set on an outbound SIP channel prior to dialing.
+
+
+
+ types of media offered
+
+
+
+ Returns the codecs offered based upon the media choice
+
+
+
+
+
+ R/O Retrieve media related information.
+
+ When rtp is specified, the
+ type parameter must be provided. It specifies
+ which RTP parameter to read.
+
+
+ Retrieve the local address for RTP.
+
+
+ Retrieve the remote address for RTP.
+
+
+ If direct media is enabled, this address is the remote address
+ used for RTP.
+
+
+ Whether or not the media stream is encrypted.
+
+
+ The media stream is not encrypted.
+
+
+ The media stream is encrypted.
+
+
+
+
+ Whether or not the media stream is currently restricted
+ due to a call hold.
+
+
+ The media stream is not held.
+
+
+ The media stream is held.
+
+
+
+
+
+
+ When rtp is specified, the
+ media_type parameter may be provided. It specifies
+ which media stream the chosen RTP parameter should be retrieved
+ from.
+
+
+ Retrieve information from the audio media stream.
+ If not specified, audio is used
+ by default.
+
+
+ Retrieve information from the video media stream.
+
+
+
+
+
+ R/O Retrieve RTCP statistics.
+
+ When rtcp is specified, the
+ statistic parameter must be provided. It specifies
+ which RTCP statistic parameter to read.
+
+
+ Retrieve a summary of all RTCP statistics.
+ The following data items are returned in a semi-colon
+ delineated list:
+
+
+ Our Synchronization Source identifier
+
+
+ Their Synchronization Source identifier
+
+
+ Our lost packet count
+
+
+ Received packet jitter
+
+
+ Received packet count
+
+
+ Transmitted packet jitter
+
+
+ Transmitted packet count
+
+
+ Remote lost packet count
+
+
+ Round trip time
+
+
+
+
+ Retrieve a summary of all RTCP Jitter statistics.
+ The following data items are returned in a semi-colon
+ delineated list:
+
+
+ Our minimum jitter
+
+
+ Our max jitter
+
+
+ Our average jitter
+
+
+ Our jitter standard deviation
+
+
+ Their minimum jitter
+
+
+ Their max jitter
+
+
+ Their average jitter
+
+
+ Their jitter standard deviation
+
+
+
+
+ Retrieve a summary of all RTCP packet loss statistics.
+ The following data items are returned in a semi-colon
+ delineated list:
+
+
+ Our minimum lost packets
+
+
+ Our max lost packets
+
+
+ Our average lost packets
+
+
+ Our lost packets standard deviation
+
+
+ Their minimum lost packets
+
+
+ Their max lost packets
+
+
+ Their average lost packets
+
+
+ Their lost packets standard deviation
+
+
+
+
+ Retrieve a summary of all RTCP round trip time information.
+ The following data items are returned in a semi-colon
+ delineated list:
+
+
+ Minimum round trip time
+
+
+ Maximum round trip time
+
+
+ Average round trip time
+
+
+ Standard deviation round trip time
+
+
+
+ Transmitted packet count
+ Received packet count
+ Transmitted packet jitter
+ Received packet jitter
+ Their max jitter
+ Their minimum jitter
+ Their average jitter
+ Their jitter standard deviation
+ Our max jitter
+ Our minimum jitter
+ Our average jitter
+ Our jitter standard deviation
+ Transmitted packet loss
+ Received packet loss
+ Their max lost packets
+ Their minimum lost packets
+ Their average lost packets
+ Their lost packets standard deviation
+ Our max lost packets
+ Our minimum lost packets
+ Our average lost packets
+ Our lost packets standard deviation
+ Round trip time
+ Maximum round trip time
+ Minimum round trip time
+ Average round trip time
+ Standard deviation round trip time
+ Our Synchronization Source identifier
+ Their Synchronization Source identifier
+
+
+
+ When rtcp is specified, the
+ media_type parameter may be provided. It specifies
+ which media stream the chosen RTCP parameter should be retrieved
+ from.
+
+
+ Retrieve information from the audio media stream.
+ If not specified, audio is used
+ by default.
+
+
+ Retrieve information from the video media stream.
+
+
+
+
+
+ R/O The name of the endpoint associated with this channel.
+ Use the PJSIP_ENDPOINT function to obtain
+ further endpoint related information.
+
+
+ R/O Obtain information about the current PJSIP channel and its
+ session.
+
+ When pjsip is specified, the
+ type parameter must be provided. It specifies
+ which signalling parameter to read.
+
+
+ Whether or not the signalling uses a secure transport.
+
+ The signalling uses a non-secure transport.
+ The signalling uses a secure transport.
+
+
+
+ The request URI of the INVITE request associated with the creation of this channel.
+
+
+ The local URI.
+
+
+ The remote URI.
+
+
+ The current state of any T.38 fax on this channel.
+
+ T.38 faxing is disabled on this channel.
+ Asterisk has sent a re-INVITE to the remote end to initiate a T.38 fax.
+ The remote end has sent a re-INVITE to Asterisk to initiate a T.38 fax.
+ A T.38 fax session has been enabled.
+ A T.38 fax session was attempted but was rejected.
+
+
+
+ On inbound calls, the full IP address and port number that
+ the INVITE request was received on. On outbound
+ calls, the full IP address and port number that the INVITE
+ request was transmitted from.
+
+
+ On inbound calls, the full IP address and port number that
+ the INVITE request was received from. On outbound
+ calls, the full IP address and port number that the INVITE
+ request was transmitted to.
+
+
+
+
+
+
+***/
+
+#include "asterisk.h"
+
+#include
+#include
+#include
+
+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 0000000000..b229a0487f
--- /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 0000000000..cbc06f0761
--- /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
diff --git a/funcs/func_channel.c b/funcs/func_channel.c
index ccdbb6e7d3..af9a6a9843 100644
--- a/funcs/func_channel.c
+++ b/funcs/func_channel.c
@@ -280,6 +280,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
Defaults to audio if unspecified.
+
chan_iax2 provides the following additional options:
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index 7b359cdf1f..615a6214cc 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -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;
diff --git a/main/xmldoc.c b/main/xmldoc.c
index 8a54d6f120..7a48c87015 100644
--- a/main/xmldoc.c
+++ b/main/xmldoc.c
@@ -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);
@@ -1490,58 +1491,6 @@ static int xmldoc_parse_specialtags(struct ast_xml_node *fixnode, const char *ta
return ret;
}
-/*!
- * \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 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, "%sTechnology: %s\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 element from the xml documentation.
@@ -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);
@@ -2051,6 +2001,60 @@ static void xmldoc_parse_parameter(struct ast_xml_node *fixnode, const char *tab
ast_free(internaltabs);
}
+/*!
+ * \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 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, "%sTechnology: %s\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
diff --git a/res/res_pjsip_t38.c b/res/res_pjsip_t38.c
index 3c97784a13..afe12505d2 100644
--- a/res/res_pjsip_t38.c
+++ b/res/res_pjsip_t38.c
@@ -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) {