[svn-commits] file: branch file/pimp_sip_location r381819 - in /team/file/pimp_sip_location...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Feb 20 08:37:11 CST 2013
Author: file
Date: Wed Feb 20 08:37:07 2013
New Revision: 381819
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=381819
Log:
Resurrect request user overwriting and add a dialplan function which calls all contacts on an AOR.
Modified:
team/file/pimp_sip_location/channels/chan_gulp.c
team/file/pimp_sip_location/res/res_sip_session.c
Modified: team/file/pimp_sip_location/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_location/channels/chan_gulp.c?view=diff&rev=381819&r1=381818&r2=381819
==============================================================================
--- team/file/pimp_sip_location/channels/chan_gulp.c (original)
+++ team/file/pimp_sip_location/channels/chan_gulp.c Wed Feb 20 08:37:07 2013
@@ -56,6 +56,28 @@
#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";
@@ -105,6 +127,90 @@
.session_end = gulp_session_end,
.incoming_request = gulp_incoming_request,
.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, ","))) {
+ struct ast_sip_aor *aor = ast_sip_location_retrieve_aor(aor_name);
+ struct ao2_container *contacts;
+ 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 a well */
+ ao2_cleanup(aor);
+ continue;
+ } else if (!ao2_container_count(contacts)) {
+ /* We were given a container but no contacts are in it... */
+ ao2_cleanup(contacts);
+ ao2_cleanup(aor);
+ 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);
+
+ ao2_cleanup(contacts);
+ ao2_cleanup(aor);
+ }
+
+ /* 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 */
@@ -923,6 +1029,11 @@
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;
@@ -931,6 +1042,8 @@
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;
@@ -946,6 +1059,7 @@
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/file/pimp_sip_location/res/res_sip_session.c
URL: http://svnview.digium.com/svn/asterisk/team/file/pimp_sip_location/res/res_sip_session.c?view=diff&rev=381819&r1=381818&r2=381819
==============================================================================
--- team/file/pimp_sip_location/res/res_sip_session.c (original)
+++ team/file/pimp_sip_location/res/res_sip_session.c Wed Feb 20 08:37:07 2013
@@ -554,14 +554,14 @@
return session;
}
-static int sip_session_create_leg(const struct ast_sip_endpoint *endpoint, struct ast_sip_session *session, const char *uri, pjsip_timer_setting *timer)
+static int sip_session_create_leg(const struct ast_sip_endpoint *endpoint, struct ast_sip_session *session, const char *uri, pjsip_timer_setting *timer, const char *request_user)
{
pjsip_dialog *dlg;
struct pjsip_inv_session *inv_session;
pjmedia_sdp_session *offer;
RAII_VAR(struct sip_session_leg *, leg, NULL, ao2_cleanup);
- if (!(dlg = ast_sip_create_dialog(endpoint, uri, NULL))) {
+ if (!(dlg = ast_sip_create_dialog(endpoint, uri, request_user))) {
return -1;
}
@@ -593,7 +593,7 @@
}
/*! \brief Internal function which adds an AOR with container of contacts as outgoing legs on a session */
-static int sip_session_leg_add_aor(struct ast_sip_endpoint *endpoint, struct ast_sip_session *session, pjsip_timer_setting *timer, const char *aor_name)
+static int sip_session_leg_add_aor(struct ast_sip_endpoint *endpoint, struct ast_sip_session *session, pjsip_timer_setting *timer, const char *aor_name, const char *request_user)
{
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);
@@ -609,7 +609,7 @@
it_contacts = ao2_iterator_init(contacts, 0);
for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
- sip_session_create_leg(endpoint, session, contact->uri, timer);
+ sip_session_create_leg(endpoint, session, contact->uri, timer, request_user);
}
ao2_iterator_destroy(&it_contacts);
@@ -644,8 +644,8 @@
while ((aor_name = strsep(&rest, ","))) {
/* If adding based on AOR fails try adding it straight up as a URI */
- if (sip_session_leg_add_aor(endpoint, session, &timer, aor_name)) {
- sip_session_create_leg(endpoint, session, aor_name, &timer);
+ if (sip_session_leg_add_aor(endpoint, session, &timer, aor_name, request_user)) {
+ sip_session_create_leg(endpoint, session, aor_name, &timer, request_user);
}
}
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