[svn-commits] mmichelson: branch mmichelson/outbound_auth r383211 - in /team/mmichelson/out...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Fri Mar 15 09:00:47 CDT 2013
Author: mmichelson
Date: Fri Mar 15 09:00:43 2013
New Revision: 383211
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=383211
Log:
Resolve conflicts and reset automerge.
Added:
team/mmichelson/outbound_auth/res/res_sip/location.c
- copied unchanged from r383209, team/group/pimp_my_sip/res/res_sip/location.c
team/mmichelson/outbound_auth/res/res_sip_nat.c
- copied unchanged from r383209, team/group/pimp_my_sip/res/res_sip_nat.c
Modified:
team/mmichelson/outbound_auth/ (props changed)
team/mmichelson/outbound_auth/channels/chan_gulp.c
team/mmichelson/outbound_auth/channels/chan_sip.c
team/mmichelson/outbound_auth/include/asterisk/res_sip.h
team/mmichelson/outbound_auth/include/asterisk/res_sip_session.h
team/mmichelson/outbound_auth/include/asterisk/stasis.h
team/mmichelson/outbound_auth/main/channel_internal_api.c
team/mmichelson/outbound_auth/main/http.c
team/mmichelson/outbound_auth/main/manager.c
team/mmichelson/outbound_auth/main/stasis.c
team/mmichelson/outbound_auth/main/stasis_cache.c
team/mmichelson/outbound_auth/main/tcptls.c
team/mmichelson/outbound_auth/res/res_sip.c
team/mmichelson/outbound_auth/res/res_sip.exports.in
team/mmichelson/outbound_auth/res/res_sip/config_transport.c
team/mmichelson/outbound_auth/res/res_sip/sip_configuration.c
team/mmichelson/outbound_auth/res/res_sip/sip_distributor.c
team/mmichelson/outbound_auth/res/res_sip_endpoint_identifier_ip.c
team/mmichelson/outbound_auth/res/res_sip_sdp_audio.c
team/mmichelson/outbound_auth/res/res_sip_session.c
team/mmichelson/outbound_auth/tests/test_stasis.c
Propchange: team/mmichelson/outbound_auth/
------------------------------------------------------------------------------
automerge = *
Propchange: team/mmichelson/outbound_auth/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.
Propchange: team/mmichelson/outbound_auth/
------------------------------------------------------------------------------
--- outbound_auth-integrated (original)
+++ outbound_auth-integrated Fri Mar 15 09:00:43 2013
@@ -1,1 +1,1 @@
-/team/group/pimp_my_sip:1-383136
+/team/group/pimp_my_sip:1-383210
Propchange: team/mmichelson/outbound_auth/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Fri Mar 15 09:00:43 2013
@@ -1,1 +1,1 @@
-/trunk:1-383127
+/trunk:1-383175
Modified: team/mmichelson/outbound_auth/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/outbound_auth/channels/chan_gulp.c?view=diff&rev=383211&r1=383210&r2=383211
==============================================================================
--- team/mmichelson/outbound_auth/channels/chan_gulp.c (original)
+++ team/mmichelson/outbound_auth/channels/chan_gulp.c Fri Mar 15 09:00:43 2013
@@ -57,8 +57,59 @@
#include "asterisk/res_sip.h"
#include "asterisk/res_sip_session.h"
+/*** DOCUMENTATION
+ <function name="GULP_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>
+ ***/
+
static const char desc[] = "Gulp SIP Channel";
static const char channel_type[] = "Gulp";
+
+/*!
+ * \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 gulp_pvt {
+ struct ast_sip_session *session;
+ struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
+};
+
+static void gulp_pvt_dtor(void *obj)
+{
+ struct gulp_pvt *pvt = obj;
+ int i;
+ ao2_cleanup(pvt->session);
+ pvt->session = NULL;
+ for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
+ ao2_cleanup(pvt->media[i]);
+ pvt->media[i] = NULL;
+ }
+}
/* \brief Asterisk core interaction functions */
static struct ast_channel *gulp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
@@ -107,17 +158,95 @@
.incoming_response = gulp_incoming_response,
};
+/*! \brief Dialplan function for constructing a dial string for calling all contacts */
+static int gulp_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+{
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(endpoint_name);
+ AST_APP_ARG(aor_name);
+ AST_APP_ARG(request_user);
+ );
+ RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
+ const char *aor_name;
+ char *rest;
+ RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
+
+ 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, "Gulp/");
+
+ 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 gulp_dial_contacts_function = {
+ .name = "GULP_DIAL_CONTACTS",
+ .read = gulp_dial_contacts,
+};
+
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result gulp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(chan);
-
- if (!session || !session->media[AST_SIP_MEDIA_AUDIO].rtp) {
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
+
+ if (!pvt || !pvt->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
return AST_RTP_GLUE_RESULT_FORBID;
}
- ao2_ref(session->media[AST_SIP_MEDIA_AUDIO].rtp, +1);
- *instance = session->media[AST_SIP_MEDIA_AUDIO].rtp;
+ *instance = pvt->media[SIP_MEDIA_AUDIO]->rtp;
+ ao2_ref(*instance, +1);
return AST_RTP_GLUE_RESULT_LOCAL;
}
@@ -146,16 +275,28 @@
{
struct ast_channel *chan;
struct ast_format fmt;
+ struct gulp_pvt *pvt;
+
+ if (!(pvt = ao2_alloc(sizeof(*pvt), gulp_pvt_dtor))) {
+ return NULL;
+ }
if (!(chan = ast_channel_alloc(1, state, "", S_OR(cid_name, ""), "", "", "", linkedid, 0, "Gulp/%s-%.*s", ast_sorcery_object_get_id(session->endpoint),
(int)session->inv_session->dlg->call_id->id.slen, session->inv_session->dlg->call_id->id.ptr))) {
+ ao2_cleanup(pvt);
return NULL;
}
ast_channel_tech_set(chan, &gulp_tech);
ao2_ref(session, +1);
- ast_channel_tech_pvt_set(chan, session);
+ pvt->session = session;
+ /* If res_sip_session is ever updated to create/destroy ast_sip_session_media
+ * during a call such as if multiple same-type stream support is introduced,
+ * these will need to be recaptured as well */
+ pvt->media[SIP_MEDIA_AUDIO] = ao2_find(session->media, "audio", OBJ_KEY);
+ pvt->media[SIP_MEDIA_VIDEO] = ao2_find(session->media, "video", OBJ_KEY);
+ ast_channel_tech_pvt_set(chan, pvt);
ast_format_cap_copy(ast_channel_nativeformats(chan), session->endpoint->codecs);
ast_codec_choose(&session->endpoint->prefs, session->endpoint->codecs, 1, &fmt);
@@ -195,7 +336,8 @@
/*! \brief Function called by core when we should answer a Gulp session */
static int gulp_answer(struct ast_channel *ast)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(ast);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
+ struct ast_sip_session *session = pvt->session;
if (ast_channel_state(ast) == AST_STATE_UP) {
return 0;
@@ -215,15 +357,20 @@
/*! \brief Function called by core to read any waiting frames */
static struct ast_frame *gulp_read(struct ast_channel *ast)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(ast);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
struct ast_frame *f;
+ struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
+
+ if (!media) {
+ return &ast_null_frame;
+ }
switch (ast_channel_fdno(ast)) {
case 0:
- f = ast_rtp_instance_read(session->media[AST_SIP_MEDIA_AUDIO].rtp, 0);
+ f = ast_rtp_instance_read(media->rtp, 0);
break;
case 1:
- f = ast_rtp_instance_read(session->media[AST_SIP_MEDIA_AUDIO].rtp, 1);
+ f = ast_rtp_instance_read(media->rtp, 1);
break;
default:
f = &ast_null_frame;
@@ -244,11 +391,17 @@
/*! \brief Function called by core to write frames */
static int gulp_write(struct ast_channel *ast, struct ast_frame *frame)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(ast);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
int res = 0;
+ struct ast_sip_session_media *media;
switch (frame->frametype) {
case AST_FRAME_VOICE:
+ media = pvt->media[SIP_MEDIA_AUDIO];
+
+ if (!media) {
+ return 0;
+ }
if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
char buf[256];
@@ -260,8 +413,8 @@
ast_getformatname(ast_channel_writeformat(ast)));
return 0;
}
- if (session->media[AST_SIP_MEDIA_AUDIO].rtp) {
- res = ast_rtp_instance_write(session->media[AST_SIP_MEDIA_AUDIO].rtp, frame);
+ if (media->rtp) {
+ res = ast_rtp_instance_write(media->rtp, frame);
}
break;
default:
@@ -287,7 +440,8 @@
/*! \brief Function called by core to change the underlying owner channel */
static int gulp_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(newchan);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(newchan);
+ struct ast_sip_session *session = pvt->session;
struct fixup_data fix_data;
fix_data.session = session;
fix_data.chan = newchan;
@@ -359,7 +513,8 @@
static int gulp_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
{
int res = 0;
- struct ast_sip_session *session = ast_channel_tech_pvt(ast);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
+ struct ast_sip_session *session = pvt->session;
int response_code = 0;
switch (condition) {
@@ -448,14 +603,18 @@
/*! \brief Function called by core to start a DTMF digit */
static int gulp_digit_begin(struct ast_channel *chan, char digit)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(chan);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(chan);
+ struct ast_sip_session *session = pvt->session;
int res = 0;
+ struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
switch (session->endpoint->dtmf) {
case AST_SIP_DTMF_RFC_4733:
- if (session->media[AST_SIP_MEDIA_AUDIO].rtp) {
- ast_rtp_instance_dtmf_begin(session->media[AST_SIP_MEDIA_AUDIO].rtp, digit);
- }
+ if (!media || !media->rtp) {
+ return -1;
+ }
+
+ ast_rtp_instance_dtmf_begin(media->rtp, digit);
case AST_SIP_DTMF_NONE:
break;
case AST_SIP_DTMF_INBAND:
@@ -471,17 +630,21 @@
/*! \brief Function called by core to stop a DTMF digit */
static int gulp_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(ast);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
+ struct ast_sip_session *session = pvt->session;
int res = 0;
+ struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
switch (session->endpoint->dtmf) {
case AST_SIP_DTMF_INFO:
/* TODO: Send INFO dtmf here */
break;
case AST_SIP_DTMF_RFC_4733:
- if (session->media[AST_SIP_MEDIA_AUDIO].rtp) {
- ast_rtp_instance_dtmf_end_with_duration(session->media[AST_SIP_MEDIA_AUDIO].rtp, digit, duration);
- }
+ if (!media || !media->rtp) {
+ return -1;
+ }
+
+ ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
case AST_SIP_DTMF_NONE:
break;
case AST_SIP_DTMF_INBAND:
@@ -510,7 +673,8 @@
/*! \brief Function called by core to actually start calling a remote party */
static int gulp_call(struct ast_channel *ast, const char *dest, int timeout)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(ast);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
+ struct ast_sip_session *session = pvt->session;
ao2_ref(session, +1);
if (ast_sip_push_task_synchronous(session->serializer, call, session)) {
@@ -597,7 +761,8 @@
pjsip_tx_data *packet = NULL;
struct hangup_data *h_data = data;
struct ast_channel *ast = h_data->chan;
- struct ast_sip_session *session = ast_channel_tech_pvt(ast);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
+ struct ast_sip_session *session = pvt->session;
int cause = h_data->cause;
if (((status = pjsip_inv_end_session(session->inv_session, cause ? cause : 603, NULL, &packet)) == PJ_SUCCESS) && packet) {
@@ -611,7 +776,7 @@
session->channel = NULL;
ast_channel_tech_pvt_set(ast, NULL);
- ao2_cleanup(session);
+ ao2_cleanup(pvt);
ao2_cleanup(h_data);
return 0;
}
@@ -619,7 +784,8 @@
/*! \brief Function called by core to hang up a Gulp session */
static int gulp_hangup(struct ast_channel *ast)
{
- struct ast_sip_session *session = ast_channel_tech_pvt(ast);
+ struct gulp_pvt *pvt = ast_channel_tech_pvt(ast);
+ struct ast_sip_session *session = pvt->session;
int cause = hangup_cause2sip(ast_channel_hangupcause(session->channel));
struct hangup_data *h_data = hangup_data_alloc(cause, ast);
if (!h_data) {
@@ -640,35 +806,56 @@
session->channel = NULL;
ast_channel_tech_pvt_set(ast, NULL);
- ao2_cleanup(session);
+ ao2_cleanup(pvt);
return -1;
}
struct request_data {
struct ast_sip_session *session;
const char *dest;
+ int cause;
};
static int request(void *obj)
{
struct request_data *req_data = obj;
- RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sip_endpoint_alloc("constant"), ao2_cleanup);
+ char *tmp = ast_strdupa(req_data->dest), *endpoint_name = NULL, *request_user = NULL;
+ RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
struct ast_sip_session *session = NULL;
-
- if (!endpoint) {
- return -1;
- }
-
- /* TODO: This needs to actually grab a proper endpoint and such */
- ast_string_field_set(endpoint, context, "default");
- ast_parse_allow_disallow(&endpoint->prefs, endpoint->codecs, "ulaw", 1);
- endpoint->min_se = 90;
- endpoint->sess_expires = 1800;
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(endpoint);
+ AST_APP_ARG(aor);
+ );
+
+ if (ast_strlen_zero(tmp)) {
+ ast_log(LOG_ERROR, "Unable to create Gulp channel with empty destination\n");
+ req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
+ return -1;
+ }
+
+ AST_NONSTANDARD_APP_ARGS(args, tmp, '/');
endpoint->sip_outbound_auths = ast_calloc(1, sizeof(char *));
endpoint->sip_outbound_auths[0] = ast_strdup("bob-auth");
endpoint->num_outbound_auths = 1;
- if (!(session = ast_sip_session_create_outgoing(endpoint, req_data->dest))) {
+ /* If a request user has been specified extract it from the endpoint name portion */
+ if ((endpoint_name = strchr(args.endpoint, '@'))) {
+ request_user = args.endpoint;
+ *endpoint_name++ = '\0';
+ } else {
+ endpoint_name = args.endpoint;
+ }
+
+ if (ast_strlen_zero(endpoint_name)) {
+ ast_log(LOG_ERROR, "Unable to create Gulp channel with empty endpoint name\n");
+ req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
+ } else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
+ ast_log(LOG_ERROR, "Unable to create Gulp channel - endpoint '%s' was not found\n", endpoint_name);
+ req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
+ return -1;
+ }
+
+ if (!(session = ast_sip_session_create_outgoing(endpoint, args.aor, request_user))) {
return -1;
}
@@ -685,6 +872,7 @@
req_data.dest = data;
if (ast_sip_push_task_synchronous(NULL, request, &req_data)) {
+ *cause = req_data.cause;
return NULL;
}
@@ -860,6 +1048,10 @@
{
struct pjsip_status_line status = rdata->msg_info.msg->line.status;
+ if (!session->channel) {
+ return;
+ }
+
switch (status.code) {
case 180:
ast_queue_control(session->channel, AST_CONTROL_RINGING);
@@ -903,6 +1095,11 @@
goto end;
}
+ if (ast_custom_function_register(&gulp_dial_contacts_function)) {
+ ast_log(LOG_ERROR, "Unable to register GULP_DIAL_CONTACTS dialplan function\n");
+ goto end;
+ }
+
if (ast_sip_session_register_supplement(&gulp_supplement)) {
ast_log(LOG_ERROR, "Unable to register Gulp supplement\n");
goto end;
@@ -911,6 +1108,8 @@
return 0;
end:
+ ast_custom_function_unregister(&gulp_dial_contacts_function);
+ ast_channel_unregister(&gulp_tech);
ast_rtp_glue_unregister(&gulp_rtp_glue);
return AST_MODULE_LOAD_FAILURE;
@@ -926,6 +1125,7 @@
static int unload_module(void)
{
ast_sip_session_unregister_supplement(&gulp_supplement);
+ ast_custom_function_unregister(&gulp_dial_contacts_function);
ast_channel_unregister(&gulp_tech);
ast_rtp_glue_unregister(&gulp_rtp_glue);
Modified: team/mmichelson/outbound_auth/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/outbound_auth/channels/chan_sip.c?view=diff&rev=383211&r1=383210&r2=383211
==============================================================================
--- team/mmichelson/outbound_auth/channels/chan_sip.c (original)
+++ team/mmichelson/outbound_auth/channels/chan_sip.c Fri Mar 15 09:00:43 2013
@@ -31646,8 +31646,11 @@
continue;
}
- /* handle tls conf */
- if (!ast_tls_read_conf(&default_tls_cfg, &sip_tls_desc, v->name, v->value)) {
+ /* handle tls conf, don't allow setting of tlsverifyclient as it isn't supported by chan_sip */
+ if (!strcasecmp(v->name, "tlsverifyclient")) {
+ ast_log(LOG_WARNING, "Ignoring unsupported option 'tlsverifyclient'\n");
+ continue;
+ } else if (!ast_tls_read_conf(&default_tls_cfg, &sip_tls_desc, v->name, v->value)) {
continue;
}
Modified: team/mmichelson/outbound_auth/include/asterisk/res_sip.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/outbound_auth/include/asterisk/res_sip.h?view=diff&rev=383211&r1=383210&r2=383211
==============================================================================
--- team/mmichelson/outbound_auth/include/asterisk/res_sip.h (original)
+++ team/mmichelson/outbound_auth/include/asterisk/res_sip.h Fri Mar 15 09:00:43 2013
@@ -28,6 +28,8 @@
#include "asterisk/channel.h"
/* Needed for ast_sorcery */
#include "asterisk/sorcery.h"
+/* Needed for ast_dnsmgr */
+#include "asterisk/dnsmgr.h"
/* Needed for pj_sockaddr */
#include <pjlib.h>
@@ -39,6 +41,7 @@
struct pjsip_transport;
struct pjsip_tpfactory;
struct pjsip_tls_setting;
+struct pjsip_tpselector;
/*!
* \brief Structure for SIP transport information
@@ -91,6 +94,10 @@
AST_STRING_FIELD(privkey_file);
/*! Password to open the private key */
AST_STRING_FIELD(password);
+ /*! External signaling address */
+ AST_STRING_FIELD(external_signaling_address);
+ /*! External media address */
+ AST_STRING_FIELD(external_media_address);
);
/*! Type of transport */
enum ast_sip_transport_type type;
@@ -98,43 +105,58 @@
pj_sockaddr host;
/*! Number of simultaneous asynchronous operations */
unsigned int async_operations;
+ /*! Optional external port for signaling */
+ unsigned int external_signaling_port;
/*! TLS settings */
pjsip_tls_setting tls;
/*! Configured TLS ciphers */
pj_ssl_cipher ciphers[SIP_TLS_MAX_CIPHERS];
+ /*! Optional local network information, used for NAT purposes */
+ struct ast_ha *localnet;
+ /*! DNS manager for refreshing the external address */
+ struct ast_dnsmgr_entry *external_address_refresher;
+ /*! Optional external address information */
+ struct ast_sockaddr external_address;
/*! Transport state information */
struct ast_sip_transport_state *state;
};
/*!
+ * \brief Structure for SIP nat hook information
+ */
+struct ast_sip_nat_hook {
+ /*! Sorcery object details */
+ SORCERY_OBJECT(details);
+ /*! Callback for when a message is going outside of our local network */
+ void (*outgoing_external_message)(struct pjsip_tx_data *tdata, struct ast_sip_transport *transport);
+};
+
+/*!
* \brief Contact associated with an address of record
*/
struct ast_sip_contact {
+ /*! Sorcery object details, the id is the aor name plus a random string */
+ SORCERY_OBJECT(details);
AST_DECLARE_STRING_FIELDS(
- /* XXX This may work better as an object instead of a string */
/*! Full URI of the contact */
AST_STRING_FIELD(uri);
);
- /*! Absolute time that this contact expires */
- time_t expiration_time;
- /*! Next list item */
- AST_LIST_ENTRY(ast_sip_contact) next;
+ /*! Absolute time that this contact is no longer valid after */
+ struct timeval expiration_time;
};
/*!
* \brief A SIP address of record
*/
struct ast_sip_aor {
- AST_DECLARE_STRING_FIELDS(
- /*! Name of the address of record */
- AST_STRING_FIELD(name);
- );
+ /*! Sorcery object details, the id is the AOR name */
+ SORCERY_OBJECT(details);
/*! Default contact expiration if one is not provided in the contact */
- int expiration;
- /*! Domain for the AOR */
- struct ast_sip_domain *domain;
- /*! Contacts bound to this AOR */
- AST_LIST_HEAD_NOLOCK(, ast_sip_contact) contacts;
+ unsigned int default_expiration;
+ /*! Maximum number of external contacts, 0 to disable */
+ unsigned int max_contacts;
+ /*! Any permanent configured contacts */
+ struct ao2_container *permanent_contacts;
};
/*!
@@ -223,8 +245,14 @@
AST_STRING_FIELD(context);
/*! Name of an explicit transport to use */
AST_STRING_FIELD(transport);
+ /*! Outbound proxy to use */
+ AST_STRING_FIELD(outbound_proxy);
+ /*! Explicit AORs to dial if none are specified */
+ AST_STRING_FIELD(aors);
/*! Musiconhold class to suggest that the other side use when placing on hold */
AST_STRING_FIELD(mohsuggest);
+ /*! Optional external media address to use in SDP */
+ AST_STRING_FIELD(external_media_address);
);
/*! Identification information for this endpoint */
struct ast_party_id id;
@@ -250,8 +278,14 @@
unsigned int rtp_ipv6;
/*! Whether symmetric RTP is enabled or not */
unsigned int rtp_symmetric;
+ /*! Whether ICE support is enabled or not */
+ unsigned int ice_support;
/*! Whether to use the "ptime" attribute received from the endpoint or not */
unsigned int use_ptime;
+ /*! Whether to force using the source IP address/port for sending responses */
+ unsigned int force_rport;
+ /*! Whether to rewrite the Contact header with the source IP address/port or not */
+ unsigned int rewrite_contact;
/*! Enabled SIP extensions */
unsigned int extensions;
/*! Minimum session expiration period, in seconds */
@@ -266,31 +300,6 @@
enum ast_sip_endpoint_identifier_type ident_method;
};
-/*!
- * \brief Given an endpoint, get its IP address
- *
- * \param endpoint The endpoint whose location is desired
- * \param[out] location_buf The IP address and port of the endpoint, as a string
- * \param size The size of the location buffer
- * \return void
- *
- * XXX The usefulness of retrieving an IP address is limited. It's more
- * useful to get a hostname or FQDN so that the location can be resolved.
- */
-void ast_sip_endpoint_get_location(const struct ast_sip_endpoint *endpoint, char *location_buf, size_t size);
-
-/*!
- * \brief Given an IP address, get the SIP endpoint that is located there.
- *
- * The returned endpoint will need to have its reference count decremented with ao2_ref()
- * once the caller has finished using it.
- *
- * \param addr The IP address
- * \retval NULL Could not find an endpoint with this IP address
- * \retval non-NULL The endpoint with this IP address
- */
-struct ast_sip_endpoint *ast_sip_get_endpoint_from_location(const char *addr);
-
/*!
* \brief Possible returns from ast_sip_check_authentication
*/
@@ -466,6 +475,78 @@
int ast_sip_initialize_sorcery_transport(struct ast_sorcery *sorcery);
/*!
+ * \brief Initialize location support on a sorcery instance
+ *
+ * \param sorcery The sorcery instance
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_initialize_sorcery_location(struct ast_sorcery *sorcery);
+
+/*!
+ * \brief Retrieve a named AOR
+ *
+ * \param aor_name Name of the AOR
+ *
+ * \retval NULL if not found
+ * \retval non-NULL if found
+ */
+struct ast_sip_aor *ast_sip_location_retrieve_aor(const char *aor_name);
+
+/*!
+ * \brief Retrieve all contacts currently available for an AOR
+ *
+ * \param aor Pointer to the AOR
+ *
+ * \param NULL if no contacts available
+ * \param non-NULL if contacts available
+ */
+struct ao2_container *ast_sip_location_retrieve_aor_contacts(const struct ast_sip_aor *aor);
+
+/*!
+ * \brief Retrieve a named contact
+ *
+ * \param contact_name Name of the contact
+ *
+ * \retval NULL if not found
+ * \retval non-NULL if found
+ */
+struct ast_sip_contact *ast_sip_location_retrieve_contact(const char *contact_name);
+
+/*!
+ * \brief Add a new contact to an AOR
+ *
+ * \param aor Pointer to the AOR
+ * \param uri Full contact URI
+ * \param expiration_time Optional expiration time of the contact
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri, struct timeval expiration_time);
+
+/*!
+ * \brief Update a contact
+ *
+ * \param contact New contact object with details
+ *
+ * \retval -1 failure
+ * \retval 0 success
+ */
+int ast_sip_location_update_contact(struct ast_sip_contact *contact);
+
+/*!
+* \brief Delete a contact
+*
+* \param contact Contact object to delete
+*
+* \retval -1 failure
+* \retval 0 success
+*/
+int ast_sip_location_delete_contact(struct ast_sip_contact *contact);
+
+/*!
* \brief Initialize authentication support on a sorcery instance
*
* \param sorcery The sorcery instance
@@ -636,6 +717,18 @@
};
/*!
+ * \brief General purpose method for creating a dialog with an endpoint
+ *
+ * \param endpoint A pointer to the endpoint
+ * \param aor_name Optional name of the AOR to target, may even be an explicit SIP URI
+ * \param request_user Optional user to place into the target URI
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ */
+ pjsip_dialog *ast_sip_create_dialog(const struct ast_sip_endpoint *endpoint, const char *aor_name, const char *request_user);
+
+/*!
* \brief General purpose method for sending a SIP request
*
* Its typical use would be to send one-off messages such as an out of dialog
Modified: team/mmichelson/outbound_auth/include/asterisk/res_sip_session.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/outbound_auth/include/asterisk/res_sip_session.h?view=diff&rev=383211&r1=383210&r2=383211
==============================================================================
--- team/mmichelson/outbound_auth/include/asterisk/res_sip_session.h (original)
+++ team/mmichelson/outbound_auth/include/asterisk/res_sip_session.h Fri Mar 15 09:00:43 2013
@@ -25,6 +25,7 @@
/* Forward declarations */
struct ast_sip_endpoint;
+struct ast_sip_transport;
struct pjsip_inv_session;
struct ast_channel;
struct ast_datastore;
@@ -37,17 +38,7 @@
struct pjmedia_sdp_session;
struct ast_rtp_instance;
-/*!
- * \brief Positions of various media
- */
-enum ast_sip_session_media_position {
- /*! \brief First is audio */
- AST_SIP_MEDIA_AUDIO = 0,
- /*! \brief Second is video */
- AST_SIP_MEDIA_VIDEO,
- /*! \brief Last is the size for media details */
- AST_SIP_MEDIA_SIZE,
-};
+struct ast_sip_session_sdp_handler;
/*!
* \brief A structure containing SIP session media information
@@ -55,8 +46,12 @@
struct ast_sip_session_media {
/*! \brief RTP instance itself */
struct ast_rtp_instance *rtp;
+ /*! \brief SDP handler that setup the RTP */
+ struct ast_sip_session_sdp_handler *handler;
/*! \brief Stream is on hold */
unsigned int held:1;
+ /*! \brief Stream type this session media handles */
+ char stream_type[1];
};
/*!
@@ -81,7 +76,7 @@
/* Datastores added to the session by supplements to the session */
struct ao2_container *datastores;
/* Media streams */
- struct ast_sip_session_media media[AST_SIP_MEDIA_SIZE];
+ struct ao2_container *media;
/* Serializer for tasks relating to this SIP session */
struct ast_taskprocessor *serializer;
};
@@ -166,34 +161,45 @@
/*!
* \brief Set session details based on a stream in an incoming SDP offer or answer
* \param session The session for which the media is being negotiated
+ * \param session_media The media to be setup for this session
* \param sdp The entire SDP. Useful for getting "global" information, such as connections or attributes
* \param stream The stream on which to operate
* \retval 0 The stream was not handled by this handler. If there are other registered handlers for this stream type, they will be called.
* \retval <0 There was an error encountered. No further operation will take place and the current negotiation will be abandoned.
* \retval >0 The stream was handled by this handler. No further handler of this stream type will be called.
*/
- int (*negotiate_incoming_sdp_stream)(struct ast_sip_session *session, const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream);
+ int (*negotiate_incoming_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream);
/*!
* \brief Create an SDP media stream and add it to the outgoing SDP offer or answer
* \param session The session for which media is being added
+ * \param session_media The media to be setup for this session
* \param stream The stream on which to operate
* \retval 0 The stream was not handled by this handler. If there are other registered handlers for this stream type, they will be called.
* \retval <0 There was an error encountered. No further operation will take place and the current negotiation will be abandoned.
* \retval >0 The stream was handled by this handler. No further handler of this stream type will be called.
*/
- int (*handle_incoming_sdp_stream)(struct ast_sip_session *session, const struct pjmedia_sdp_session *sdp, struct pjmedia_sdp_media *stream);
+ int (*handle_incoming_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *sdp, struct pjmedia_sdp_media *stream);
/*!
* \brief Create an SDP media stream and add it to the outgoing SDP offer or answer
* \param session The session for which media is being added
+ * \param session_media The media to be setup for this session
* \param sdp The entire SDP as currently built
* \retval 0 This handler has no stream to add. If there are other registered handlers for this stream type, they will be called.
* \retval <0 There was an error encountered. No further operation will take place and the current SDP negotiation will be abandoned.
* \retval >0 The handler has a stream to be added to the SDP. No further handler of this stream type will be called.
*/
- int (*create_outgoing_sdp_stream)(struct ast_sip_session *session, struct pjmedia_sdp_session *sdp);
+ int (*create_outgoing_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, struct pjmedia_sdp_session *sdp);
+ /*!
+ * \brief Update media stream with external address if applicable
+ * \param tdata The outgoing message itself
+ * \param stream The stream on which to operate
+ * \param transport The transport the SDP is going out on
+ */
+ void (*change_outgoing_sdp_stream_media_address)(struct pjsip_tx_data *tdata, struct pjmedia_sdp_media *stream, struct ast_sip_transport *transport);
/*!
* \brief Apply a negotiated SDP media stream
* \param session The session for which media is being applied
+ * \param session_media The media to be setup for this session
* \param local The entire local negotiated SDP
* \param local_stream The local stream which to apply
* \param remote The entire remote negotiated SDP
@@ -202,9 +208,15 @@
* \retval <0 There was an error encountered. No further operation will take place and the current application will be abandoned.
* \retval >0 The stream was handled by this handler. No further handler of this stream type will be called.
*/
- int (*apply_negotiated_sdp_stream)(struct ast_sip_session *session, const struct pjmedia_sdp_session *local, const struct pjmedia_sdp_media *local_stream,
+ int (*apply_negotiated_sdp_stream)(struct ast_sip_session *session, struct ast_sip_session_media *session_media, const struct pjmedia_sdp_session *local, const struct pjmedia_sdp_media *local_stream,
const struct pjmedia_sdp_session *remote, const struct pjmedia_sdp_media *remote_stream);
- /*! Next item int he list. */
+ /*!
+ * \brief Destroy a session_media created by this handler
+ * \param session The session for which media is being destroyed
+ * \param session_media The media to destroy
+ */
+ void (*stream_destroy)(struct ast_sip_session_media *session_media);
+ /*! Next item in the list. */
AST_LIST_ENTRY(ast_sip_session_sdp_handler) next;
};
@@ -231,9 +243,10 @@
* this reference when the session is destroyed.
*
* \param endpoint The endpoint that this session uses for settings
- * \param uri The URI to call
- */
-struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint, const char *uri);
+ * \param location Optional name of the location to call, be it named location or explicit URI
+ * \param request_user Optional request user to place in the request URI if permitted
+ */
+struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint, const char *location, const char *request_user);
/*!
* \brief Register an SDP handler
Modified: team/mmichelson/outbound_auth/include/asterisk/stasis.h
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/outbound_auth/include/asterisk/stasis.h?view=diff&rev=383211&r1=383210&r2=383211
==============================================================================
--- team/mmichelson/outbound_auth/include/asterisk/stasis.h (original)
+++ team/mmichelson/outbound_auth/include/asterisk/stasis.h Fri Mar 15 09:00:43 2013
@@ -306,9 +306,10 @@
* delivery of the final message.
*
* \param subscription Subscription to cancel.
- * \since 12
- */
-void stasis_unsubscribe(struct stasis_subscription *subscription);
+ * \retval NULL for convenience
+ * \since 12
+ */
+struct stasis_subscription *stasis_unsubscribe(struct stasis_subscription *subscription);
/*!
* \brief Create a subscription which forwards all messages from one topic to
@@ -450,9 +451,10 @@
/*!
* Unsubscribes a caching topic from its upstream topic.
* \param caching_topic Caching topic to unsubscribe
- * \since 12
- */
-void stasis_caching_unsubscribe(struct stasis_caching_topic *caching_topic);
+ * \retval NULL for convenience
+ * \since 12
+ */
+struct stasis_caching_topic *stasis_caching_unsubscribe(struct stasis_caching_topic *caching_topic);
/*!
* \brief Returns the topic of cached events from a caching topics.
Modified: team/mmichelson/outbound_auth/main/channel_internal_api.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/outbound_auth/main/channel_internal_api.c?view=diff&rev=383211&r1=383210&r2=383211
==============================================================================
--- team/mmichelson/outbound_auth/main/channel_internal_api.c (original)
+++ team/mmichelson/outbound_auth/main/channel_internal_api.c Fri Mar 15 09:00:43 2013
@@ -1367,8 +1367,7 @@
ast_string_field_free_memory(chan);
- stasis_unsubscribe(chan->forwarder);
- chan->forwarder = NULL;
+ chan->forwarder = stasis_unsubscribe(chan->forwarder);
ao2_cleanup(chan->topic);
chan->topic = NULL;
Modified: team/mmichelson/outbound_auth/main/http.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/outbound_auth/main/http.c?view=diff&rev=383211&r1=383210&r2=383211
==============================================================================
--- team/mmichelson/outbound_auth/main/http.c (original)
+++ team/mmichelson/outbound_auth/main/http.c Fri Mar 15 09:00:43 2013
@@ -1060,8 +1060,17 @@
v = ast_variable_browse(cfg, "general");
for (; v; v = v->next) {
- /* handle tls conf */
- if (!ast_tls_read_conf(&http_tls_cfg, &https_desc, v->name, v->value)) {
+ /* read tls config options while preventing unsupported options from being set */
+ if (strcasecmp(v->name, "tlscafile")
+ && strcasecmp(v->name, "tlscapath")
+ && strcasecmp(v->name, "tlscadir")
+ && strcasecmp(v->name, "tlsverifyclient")
+ && strcasecmp(v->name, "tlsdontverifyserver")
+ && strcasecmp(v->name, "tlsclientmethod")
+ && strcasecmp(v->name, "sslclientmethod")
+ && strcasecmp(v->name, "tlscipher")
+ && strcasecmp(v->name, "sslcipher")
+ && !ast_tls_read_conf(&http_tls_cfg, &https_desc, v->name, v->value)) {
continue;
}
Modified: team/mmichelson/outbound_auth/main/manager.c
URL: http://svnview.digium.com/svn/asterisk/team/mmichelson/outbound_auth/main/manager.c?view=diff&rev=383211&r1=383210&r2=383211
==============================================================================
--- team/mmichelson/outbound_auth/main/manager.c (original)
+++ team/mmichelson/outbound_auth/main/manager.c Fri Mar 15 09:00:43 2013
@@ -7590,8 +7590,7 @@
{
struct ast_manager_user *user;
- stasis_unsubscribe(channel_state_sub);
- channel_state_sub = NULL;
+ channel_state_sub = stasis_unsubscribe(channel_state_sub);
if (registered) {
ast_manager_unregister("Ping");
@@ -7795,7 +7794,15 @@
for (var = ast_variable_browse(cfg, "general"); var; var = var->next) {
val = var->value;
- if (!ast_tls_read_conf(&ami_tls_cfg, &amis_desc, var->name, val)) {
[... 2082 lines stripped ...]
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