Add a "message_context" option for PJSIP endpoints.

........

Merged revisions 411157 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

UPGRADE.txt

@@ -26,6 +26,9 @@ PJSIP:
    REGISTER requests for each contact that is registered. If using realtime for
    PJSIP contacts, this means that the schema has been updated to add a user_agent
    column. An alembic revision has been added to facilitate this update.
+ 
+ - PJSIP endpoints now have a "message_context" option that can be used to determine
+   where to route incoming MESSAGE requests from the endpoint.
 
 Realtime Configuration:
  - PJSIP endpoint columns 'tos_audio' and 'tos_video' have been changed from yes/no

include/asterisk/res_pjsip.h

@@ -562,6 +562,8 @@ struct ast_sip_endpoint {
 		AST_STRING_FIELD(fromuser);
 		/*! Domain to place in From header */
 		AST_STRING_FIELD(fromdomain);
+		/*! Context to route incoming MESSAGE requests to */
+		AST_STRING_FIELD(message_context);
 	);
 	/*! Configuration for extensions */
 	struct ast_sip_endpoint_extensions extensions;

res/res_pjsip.c

@@ -685,6 +685,14 @@
 						multiple 'set_var'(s).
 					</para></description>
 				</configOption>
+				<configOption name="message_context">
+					<synopsis>Context to route incoming MESSAGE requests to.</synopsis>
+					<description><para>
+						If specified, incoming MESSAGE requests will be routed to the indicated
+						dialplan context. If no <replaceable>message_context</replaceable> is
+						specified, then the <replaceable>context</replaceable> setting is used.
+					</para></description>
+				</configOption>
 			</configObject>
 			<configObject name="auth">
 				<synopsis>Authentication type</synopsis>

res/res_pjsip/pjsip_configuration.c

@@ -1723,6 +1723,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
 	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "srtp_tag_32", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, media.rtp.srtp_tag_32));
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "redirect_method", "user", redirect_handler, NULL, NULL, 0, 0);
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "set_var", "", set_var_handler, set_var_to_str, set_var_to_vl, 0, 0);
+	ast_sorcery_object_field_register(sip_sorcery, "endpoint", "message_context", "", OPT_STRINGFIELD_T, 1, STRFLDSET(struct ast_sip_endpoint, message_context));
 
 	if (ast_sip_initialize_sorcery_transport()) {
 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");

res/res_pjsip_messaging.c

@@ -464,13 +464,14 @@ static enum pjsip_status_code rx_data_to_ast_msg(pjsip_rx_data *rdata, struct as
 	const char *field;
 	pjsip_status_code code;
 	struct ast_sip_endpoint *endpt = ast_pjsip_rdata_get_endpoint(rdata);
+	const char *context = S_OR(endpt->message_context, endpt->context);
 
 	/* make sure there is an appropriate context and extension*/
-	if ((code = get_destination(rdata, endpt->context, buf)) != PJSIP_SC_OK) {
+	if ((code = get_destination(rdata, context, buf)) != PJSIP_SC_OK) {
 		return code;
 	}
 
-	CHECK_RES(ast_msg_set_context(msg, "%s", endpt->context));
+	CHECK_RES(ast_msg_set_context(msg, "%s", context));
 	CHECK_RES(ast_msg_set_exten(msg, "%s", buf));
 
 	/* to header */