res_pjsip: Add ignore_uri_user_options option.

This implements the chan_sip legacy_useroption_parsing option but with a
better name.

* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.

ASTERISK-26316 #close
Reported by: Kevin Harwell

Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62

CHANGES

@@ -8,6 +8,23 @@
 ===
 ==============================================================================
 
+------------------------------------------------------------------------------
+--- Functionality changes from Asterisk 13.8-cert3 to Asterisk 13.8-cert4 ----
+------------------------------------------------------------------------------
+
+res_pjsip
+------------------
+ * Added "ignore_uri_user_options" global configuration option for
+   compatibility with an ITSP that sends URI user field options.  When enabled
+   the user field is truncated at the first semicolon.
+   Example:
+   URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
+   The user field is "1235557890;phone-context=national"
+   Which is truncated to this: "1235557890"
+
+   Note: The caller-id and redirecting number strings obtained from incoming
+   SIP URI user fields are now always truncated at the first semicolon.
+
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 13.8-cert2 to Asterisk 13.8-cert3 ----
 ------------------------------------------------------------------------------

configs/samples/pjsip.conf.sample

@@ -910,6 +910,22 @@
 ; with us. The extension added is the name of the endpoint.
 ;regcontext=sipregistrations
 
+;ignore_uri_user_options=no ; Enable/Disable ignoring SIP URI user field options.
+                    ; If you have this option enabled and there are semicolons
+                    ; in the user field of a SIP URI then the field is truncated
+                    ; at the first semicolon.  This effectively makes the semicolon
+                    ; a non-usable character for PJSIP endpoint names, extensions,
+                    ; and AORs.  This can be useful for improving compatability with
+                    ; an ITSP that likes to use user options for whatever reason.
+                    ; Example:
+                    ; URI: "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
+                    ; The user field is "1235557890;phone-context=national"
+                    ; Which becomes this: "1235557890"
+                    ;
+                    ; Note: The caller-id and redirecting number strings obtained
+                    ; from incoming SIP URI user fields are always truncated at the
+                    ; first semicolon.
+
 ; MODULE PROVIDING BELOW SECTION(S): res_pjsip_acl
 ;==========================ACL SECTION OPTIONS=========================
 ;[acl]

contrib/ast-db-manage/config/versions/a6ef36f1309_ps_globals_add_ignore_uri_user_options.py

@@ -0,0 +1,32 @@
+"""ps_globals add ignore_uri_user_options
+
+Revision ID: a6ef36f1309
+Revises: 4e2493ef32e6
+Create Date: 2016-08-31 12:24:22.368956
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = 'a6ef36f1309'
+down_revision = '4e2493ef32e6'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+def upgrade():
+    ############################# Enums ##############################
+
+    # yesno_values have already been created, so use postgres enum object
+    # type to get around "already created" issue - works okay with mysql
+    yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+
+    op.add_column('ps_globals', sa.Column('ignore_uri_user_options', yesno_values))
+
+
+def downgrade():
+    op.drop_column('ps_globals', 'ignore_uri_user_options')
+

include/asterisk/res_pjsip.h

@@ -2192,6 +2192,38 @@ int ast_sip_register_supplement(struct ast_sip_supplement *supplement);
  */
 void ast_sip_unregister_supplement(struct ast_sip_supplement *supplement);
 
+/*!
+ * \brief Retrieve the global setting 'ignore_uri_user_options'.
+ * \since 13.12.0
+ *
+ * \retval non zero if ignore the user field options.
+ */
+unsigned int ast_sip_get_ignore_uri_user_options(void);
+
+/*!
+ * \brief Truncate the URI user field options string if enabled.
+ * \since 13.12.0
+ *
+ * \param str URI user field string to truncate if enabled
+ *
+ * \details
+ * We need to be able to handle URI's looking like
+ * "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
+ *
+ * Where the URI user field is:
+ * "1235557890;phone-context=national"
+ *
+ * When truncated the string will become:
+ * "1235557890"
+ */
+#define AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(str)				\
+	do {														\
+		char *__semi = strchr((str), ';');						\
+		if (__semi && ast_sip_get_ignore_uri_user_options()) {	\
+			*__semi = '\0';										\
+		}														\
+	} while (0)
+
 /*!
  * \brief Retrieve the system debug setting (yes|no|host).
  *

res/res_pjsip.c

@@ -1330,6 +1330,30 @@
                                         set to this value if there is no better option (such as CallerID) to be
                                         used.</synopsis>
 				</configOption>
+				<configOption name="ignore_uri_user_options">
+					<synopsis>Enable/Disable ignoring SIP URI user field options.</synopsis>
+					<description>
+						<para>If you have this option enabled and there are semicolons
+						in the user field of a SIP URI then the field is truncated
+						at the first semicolon.  This effectively makes the semicolon
+						a non-usable character for PJSIP endpoint names, extensions,
+						and AORs.  This can be useful for improving compatability with
+						an ITSP that likes to use user options for whatever reason.
+						</para>
+						<example title="Sample SIP URI">
+							sip:1235557890;phone-context=national@x.x.x.x;user=phone
+						</example>
+						<example title="Sample SIP URI user field">
+							1235557890;phone-context=national
+						</example>
+						<example title="Sample SIP URI user field truncated">
+							1235557890
+						</example>
+						<note><para>The caller-id and redirecting number strings
+						obtained from incoming SIP URI user fields are always truncated
+						at the first semicolon.</para></note>
+					</description>
+				</configOption>
 			</configObject>
 		</configFile>
 	</configInfo>

res/res_pjsip/config_global.c

@@ -37,6 +37,7 @@
 #define DEFAULT_FROM_USER "asterisk"
 #define DEFAULT_REGCONTEXT ""
 #define DEFAULT_DISABLE_MULTI_DOMAIN 0
+#define DEFAULT_IGNORE_URI_USER_OPTIONS 0
 
 static char default_useragent[256];
 
@@ -61,6 +62,8 @@ struct global_config {
 	unsigned int max_initial_qualify_time;
 	/*! Nonzero to disable multi domain support */
 	unsigned int disable_multi_domain;
+	/*! Nonzero if URI user field options are ignored. */
+	unsigned int ignore_uri_user_options;
 };
 
 static void global_destructor(void *obj)
@@ -232,6 +235,21 @@ void ast_sip_get_default_from_user(char *from_user, size_t size)
 	}
 }
 
+unsigned int ast_sip_get_ignore_uri_user_options(void)
+{
+	unsigned int ignore_uri_user_options;
+	struct global_config *cfg;
+
+	cfg = get_global_cfg();
+	if (!cfg) {
+		return DEFAULT_IGNORE_URI_USER_OPTIONS;
+	}
+
+	ignore_uri_user_options = cfg->ignore_uri_user_options;
+	ao2_ref(cfg, -1);
+	return ignore_uri_user_options;
+}
+
 /*!
  * \internal
  * \brief Observer to set default global object if none exist.
@@ -351,6 +369,9 @@ int ast_sip_initialize_sorcery_global(void)
 		OPT_STRINGFIELD_T, 0, STRFLDSET(struct global_config, regcontext));
 	ast_sorcery_object_field_register(sorcery, "global", "disable_multi_domain", "no",
 		OPT_BOOL_T, 1, FLDSET(struct global_config, disable_multi_domain));
+	ast_sorcery_object_field_register(sorcery, "global", "ignore_uri_user_options",
+		DEFAULT_IGNORE_URI_USER_OPTIONS ? "yes" : "no",
+		OPT_BOOL_T, 1, FLDSET(struct global_config, ignore_uri_user_options));
 
 	if (ast_sorcery_instance_observer_add(sorcery, &observer_callbacks_global)) {
 		return -1;

res/res_pjsip/pjsip_options.c

@@ -701,8 +701,7 @@ static pj_bool_t options_on_rx_request(pjsip_rx_data *rdata)
 	pjsip_sip_uri *sip_ruri;
 	char exten[AST_MAX_EXTENSION];
 
-	if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method,
-			     &pjsip_options_method)) {
+	if (pjsip_method_cmp(&rdata->msg_info.msg->line.req.method, &pjsip_options_method)) {
 		return PJ_FALSE;
 	}
 
@@ -719,13 +718,20 @@ static pj_bool_t options_on_rx_request(pjsip_rx_data *rdata)
 	sip_ruri = pjsip_uri_get_uri(ruri);
 	ast_copy_pj_str(exten, &sip_ruri->user, sizeof(exten));
 
+	/*
+	 * We may want to match in the dialplan without any user
+	 * options getting in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+
 	if (ast_shutting_down()) {
 		/*
 		 * Not taking any new calls at this time.
 		 * Likely a server availability OPTIONS poll.
 		 */
 		send_options_response(rdata, 503);
-	} else if (!ast_strlen_zero(exten) && !ast_exists_extension(NULL, endpoint->context, exten, 1, NULL)) {
+	} else if (!ast_strlen_zero(exten)
+		&& !ast_exists_extension(NULL, endpoint->context, exten, 1, NULL)) {
 		send_options_response(rdata, 404);
 	} else {
 		send_options_response(rdata, 200);

res/res_pjsip_caller_id.c

@@ -46,11 +46,29 @@ static void set_id_from_hdr(pjsip_fromto_hdr *hdr, struct ast_party_id *id)
 	char cid_num[AST_CHANNEL_NAME];
 	pjsip_sip_uri *uri;
 	pjsip_name_addr *id_name_addr = (pjsip_name_addr *) hdr->uri;
+	char *semi;
 
 	uri = pjsip_uri_get_uri(id_name_addr);
 	ast_copy_pj_str(cid_name, &id_name_addr->display, sizeof(cid_name));
 	ast_copy_pj_str(cid_num, &uri->user, sizeof(cid_num));
 
+	/* Always truncate caller-id number at a semicolon. */
+	semi = strchr(cid_num, ';');
+	if (semi) {
+		/*
+		 * We need to be able to handle URI's looking like
+		 * "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
+		 *
+		 * Where the uri->user field will result in:
+		 * "1235557890;phone-context=national"
+		 *
+		 * People don't care about anything after the semicolon
+		 * showing up on their displays even though the RFC
+		 * allows the semicolon.
+		 */
+		*semi = '\0';
+	}
+
 	ast_free(id->name.str);
 	id->name.str = ast_strdup(cid_name);
 	if (!ast_strlen_zero(cid_name)) {

res/res_pjsip_diversion.c

@@ -148,11 +148,32 @@ static void set_redirecting_id(pjsip_name_addr *name_addr, struct ast_party_id *
 			       struct ast_set_party_id *update)
 {
 	pjsip_sip_uri *uri = pjsip_uri_get_uri(name_addr->uri);
+	char *semi;
+	pj_str_t uri_user;
+
+	uri_user = uri->user;
+
+	/* Always truncate redirecting number at a semicolon. */
+	semi = pj_strchr(&uri_user, ';');
+	if (semi) {
+		/*
+		 * We need to be able to handle URI's looking like
+		 * "sip:1235557890;phone-context=national@x.x.x.x;user=phone"
+		 *
+		 * Where the uri->user field will result in:
+		 * "1235557890;phone-context=national"
+		 *
+		 * People don't care about anything after the semicolon
+		 * showing up on their displays even though the RFC
+		 * allows the semicolon.
+		 */
+		pj_strset(&uri_user, (char *) pj_strbuf(&uri_user), semi - pj_strbuf(&uri_user));
+	}
 
-	if (pj_strlen(&uri->user)) {
+	if (pj_strlen(&uri_user)) {
 		update->number = 1;
 		data->number.valid = 1;
-		set_redirecting_value(&data->number.str, &uri->user);
+		set_redirecting_value(&data->number.str, &uri_user);
 	}
 
 	if (pj_strlen(&name_addr->display)) {

res/res_pjsip_endpoint_identifier_user.c

@@ -33,6 +33,7 @@ static int get_endpoint_details(pjsip_rx_data *rdata, char *endpoint, size_t end
 {
 	pjsip_uri *from = rdata->msg_info.from->uri;
 	pjsip_sip_uri *sip_from;
+
 	if (!PJSIP_URI_SCHEME_IS_SIP(from) && !PJSIP_URI_SCHEME_IS_SIPS(from)) {
 		return -1;
 	}
@@ -69,6 +70,12 @@ static struct ast_sip_endpoint *username_identify(pjsip_rx_data *rdata)
 		return NULL;
 	}
 
+	/*
+	 * We may want to be matched without any user options getting
+	 * in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(endpoint_name);
+
 	if (!ast_sip_get_disable_multi_domain()) {
 		/* Attempt to find the endpoint given the name and domain provided */
 		snprintf(id, sizeof(id), "%s@%s", endpoint_name, domain_name);

res/res_pjsip_messaging.c

@@ -133,6 +133,12 @@ static struct ast_sip_endpoint* get_outbound_endpoint(
 	} else if ((aor_uri = strchr(name, '@'))) {
 		/* format was 'endpoint@' - don't use the rest */
 		*aor_uri = '\0';
+
+		/*
+		 * We may want to match without any user options getting
+		 * in the way.
+		 */
+		AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(name);
 	}
 
 	/* at this point, if name is not empty then it
@@ -448,6 +454,12 @@ static enum pjsip_status_code rx_data_to_ast_msg(pjsip_rx_data *rdata, struct as
 	sip_ruri = pjsip_uri_get_uri(ruri);
 	ast_copy_pj_str(exten, &sip_ruri->user, AST_MAX_EXTENSION);
 
+	/*
+	 * We may want to match in the dialplan without any user
+	 * options getting in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+
 	endpt = ast_pjsip_rdata_get_endpoint(rdata);
 	ast_assert(endpt != NULL);
 
@@ -528,7 +540,7 @@ static void msg_data_destroy(void *obj)
 
 static struct msg_data* msg_data_create(const struct ast_msg *msg, const char *to, const char *from)
 {
-	char *tag;
+	char *uri_params;
 	struct msg_data *mdata = ao2_alloc(sizeof(*mdata), msg_data_destroy);
 
 	if (!mdata) {
@@ -553,9 +565,14 @@ static struct msg_data* msg_data_create(const struct ast_msg *msg, const char *t
 		return NULL;
 	}
 
-	/* sometimes from can still contain the tag at this point, so remove it */
-	if ((tag = strchr(mdata->from, ';'))) {
-		*tag = '\0';
+	/*
+	 * Sometimes from URI can contain URI parameters, so remove them.
+	 *
+	 * sip:user;user-options@domain;uri-parameters
+	 */
+	uri_params = strchr(mdata->from, '@');
+	if (uri_params && (uri_params = strchr(mdata->from, ';'))) {
+		*uri_params = '\0';
 	}
 	return mdata;
 }

res/res_pjsip_path.c

@@ -40,7 +40,8 @@ static struct ast_sip_aor *find_aor(struct ast_sip_endpoint *endpoint, pjsip_uri
 	char *configured_aors, *aor_name;
 	pjsip_sip_uri *sip_uri;
 	char *domain_name;
-	RAII_VAR(struct ast_str *, id, NULL, ast_free);
+	char *username;
+	struct ast_str *id = NULL;
 
 	if (ast_strlen_zero(endpoint->aors)) {
 		return NULL;
@@ -49,6 +50,14 @@ static struct ast_sip_aor *find_aor(struct ast_sip_endpoint *endpoint, pjsip_uri
 	sip_uri = pjsip_uri_get_uri(uri);
 	domain_name = ast_alloca(sip_uri->host.slen + 1);
 	ast_copy_pj_str(domain_name, &sip_uri->host, sip_uri->host.slen + 1);
+	username = ast_alloca(sip_uri->user.slen + 1);
+	ast_copy_pj_str(username, &sip_uri->user, sip_uri->user.slen + 1);
+
+	/*
+	 * We may want to match without any user options getting
+	 * in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(username);
 
 	configured_aors = ast_strdupa(endpoint->aors);
 
@@ -60,15 +69,16 @@ static struct ast_sip_aor *find_aor(struct ast_sip_endpoint *endpoint, pjsip_uri
 			continue;
 		}
 
-		if (!pj_strcmp2(&sip_uri->user, aor_name)) {
+		if (!strcmp(username, aor_name)) {
 			break;
 		}
 
-		if (!id && !(id = ast_str_create(sip_uri->user.slen + sip_uri->host.slen + 2))) {
-			return NULL;
+		if (!id && !(id = ast_str_create(strlen(username) + sip_uri->host.slen + 2))) {
+			aor_name = NULL;
+			break;
 		}
 
-		ast_str_set(&id, 0, "%.*s@", (int)sip_uri->user.slen, sip_uri->user.ptr);
+		ast_str_set(&id, 0, "%s@", username);
 		if ((alias = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "domain_alias", domain_name))) {
 			ast_str_append(&id, 0, "%s", alias->domain);
 			ao2_cleanup(alias);
@@ -77,10 +87,10 @@ static struct ast_sip_aor *find_aor(struct ast_sip_endpoint *endpoint, pjsip_uri
 		}
 
 		if (!strcmp(aor_name, ast_str_buffer(id))) {
-			ast_free(id);
 			break;
 		}
 	}
+	ast_free(id);
 
 	if (ast_strlen_zero(aor_name)) {
 		return NULL;

res/res_pjsip_pubsub.c

@@ -1400,6 +1400,12 @@ static int sub_persistence_recreate(void *obj)
 	resource = ast_alloca(resource_size);
 	ast_copy_pj_str(resource, &request_uri->user, resource_size);
 
+	/*
+	 * We may want to match without any user options getting
+	 * in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(resource);
+
 	handler = subscription_get_handler_from_rdata(rdata);
 	if (!handler || !handler->notifier) {
 		ast_log(LOG_WARNING, "Failed recreating '%s' subscription: Could not get subscription handler.\n",
@@ -2796,6 +2802,12 @@ static pj_bool_t pubsub_on_rx_subscribe_request(pjsip_rx_data *rdata)
 	resource = ast_alloca(resource_size);
 	ast_copy_pj_str(resource, &request_uri_sip->user, resource_size);
 
+	/*
+	 * We may want to match without any user options getting
+	 * in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(resource);
+
 	expires_header = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_EXPIRES, rdata->msg_info.msg->hdr.next);
 
 	if (expires_header) {
@@ -3009,6 +3021,12 @@ static struct ast_sip_publication *publish_request_initial(struct ast_sip_endpoi
 	resource_name = ast_alloca(resource_size);
 	ast_copy_pj_str(resource_name, &request_uri_sip->user, resource_size);
 
+	/*
+	 * We may want to match without any user options getting
+	 * in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(resource_name);
+
 	resource = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "inbound-publication", resource_name);
 	if (!resource) {
 		pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 404, NULL, NULL, NULL);

res/res_pjsip_refer.c

@@ -809,6 +809,13 @@ static int refer_incoming_blind_request(struct ast_sip_session *session, pjsip_r
 
 	/* Using the user portion of the target URI see if it exists as a valid extension in their context */
 	ast_copy_pj_str(exten, &target->user, sizeof(exten));
+
+	/*
+	 * We may want to match in the dialplan without any user
+	 * options getting in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+
 	if (!ast_exists_extension(NULL, context, exten, 1, NULL)) {
 		ast_log(LOG_ERROR, "Channel '%s' from endpoint '%s' attempted blind transfer to '%s@%s' but target does not exist\n",
 			ast_channel_name(session->channel), ast_sorcery_object_get_id(session->endpoint), exten, context);

res/res_pjsip_registrar.c

@@ -519,6 +519,7 @@ static struct ast_sip_aor *find_registrar_aor(struct pjsip_rx_data *rdata, struc
 	struct ast_sip_aor *aor = NULL;
 	pjsip_sip_uri *uri;
 	char *domain_name;
+	char *username;
 	char *configured_aors;
 	char *aor_name;
 	struct ast_str *id = NULL;
@@ -526,6 +527,14 @@ static struct ast_sip_aor *find_registrar_aor(struct pjsip_rx_data *rdata, struc
 	uri = pjsip_uri_get_uri(rdata->msg_info.to->uri);
 	domain_name = ast_alloca(uri->host.slen + 1);
 	ast_copy_pj_str(domain_name, &uri->host, uri->host.slen + 1);
+	username = ast_alloca(uri->user.slen + 1);
+	ast_copy_pj_str(username, &uri->user, uri->user.slen + 1);
+
+	/*
+	 * We may want to match without any user options getting
+	 * in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(username);
 
 	configured_aors = ast_strdupa(endpoint->aors);
 
@@ -537,16 +546,16 @@ static struct ast_sip_aor *find_registrar_aor(struct pjsip_rx_data *rdata, struc
 			continue;
 		}
 
-		if (!pj_strcmp2(&uri->user, aor_name)) {
+		if (!strcmp(username, aor_name)) {
 			break;
 		}
 
-		if (!id && !(id = ast_str_create(uri->user.slen + uri->host.slen + 2))) {
+		if (!id && !(id = ast_str_create(strlen(username) + uri->host.slen + 2))) {
 			aor_name = NULL;
 			break;
 		}
 
-		ast_str_set(&id, 0, "%.*s@", (int)uri->user.slen, uri->user.ptr);
+		ast_str_set(&id, 0, "%s@", username);
 		if ((alias = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "domain_alias", domain_name))) {
 			ast_str_append(&id, 0, "%s", alias->domain);
 			ao2_cleanup(alias);

res/res_pjsip_session.c

@@ -1952,6 +1952,12 @@ static enum sip_get_destination_result get_destination(struct ast_sip_session *s
 	sip_ruri = pjsip_uri_get_uri(ruri);
 	ast_copy_pj_str(session->exten, &sip_ruri->user, sizeof(session->exten));
 
+	/*
+	 * We may want to match in the dialplan without any user
+	 * options getting in the way.
+	 */
+	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(session->exten);
+
 	pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
 	if (!pickup_cfg) {
 		ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
@@ -2903,6 +2909,13 @@ static pjsip_redirect_op session_inv_on_redirected(pjsip_inv_session *inv, const
 		char exten[AST_MAX_EXTENSION];
 
 		ast_copy_pj_str(exten, &uri->user, sizeof(exten));
+
+		/*
+		 * We may want to match in the dialplan without any user
+		 * options getting in the way.
+		 */
+		AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+
 		ast_channel_call_forward_set(session->channel, exten);
 	} else if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_CORE) {
 		char target_uri[PJSIP_MAX_URL_SIZE];