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Add a "message_context" option for PJSIP endpoints.

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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
This commit is contained in:
Mark Michelson
2014-03-25 17:40:51 +00:00
parent c1c8300e27
commit 2bf37a417d
5 changed files with 17 additions and 2 deletions
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3
UPGRADE.txt
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@@ -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

2
include/asterisk/res_pjsip.h
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@@ -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;

8
res/res_pjsip.c
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@@ -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>

1
res/res_pjsip/pjsip_configuration.c
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@@ -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");

5
res/res_pjsip_messaging.c
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@@ -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 */