res_pjsip: add 'set_var' support on endpoints

Added a new 'set_var' option for ast_sip_endpoint(s).  For each variable
specified that variable gets set upon creation of a pjsip channel involving
the endpoint.

(closes issue ASTERISK-22868)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/3095/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404663 65c4cc65-6c06-0410-ace0-fbb531ad65f3

CHANGES

@@ -1117,6 +1117,9 @@ res_pjsip (and many others)
    these modules is performed in pjsip.conf. Other modules may use their
    own configuration files.
 
+ * Added 'set_var' option for an endpoint. For each variable specified that
+   variable gets set upon creation of a channel involving the endpoint.
+
 res_rtp_asterisk
 ------------------
  * ICE/STUN/TURN support in res_rtp_asterisk has been made optional.  To enable

channels/chan_pjsip.c

@@ -346,6 +346,7 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
 	struct ast_format fmt;
 	RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
 	struct ast_sip_channel_pvt *channel;
+	struct ast_variable *var;
 
 	if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
 		return NULL;
@@ -364,6 +365,11 @@ static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int s
 		return NULL;
 	}
 
+	for (var = session->endpoint->channel_vars; var; var = var->next) {
+		char buf[512];
+		pbx_builtin_setvar_helper(chan, var->name, ast_get_encoded_str(
+						  var->value, buf, sizeof(buf)));
+	}
 
 	ast_channel_stage_snapshot(chan);
 

configs/pjsip.conf.sample

@@ -539,7 +539,8 @@
                 ; other party or both (default: "")
 ;srtp_tag_32=no ; Determines whether 32 byte tags should be used instead of 80
                 ; byte tags (default: "no")
-
+;set_var=       ; Variable set on a channel involving the endpoint. For multiple
+		; channel variables specify multiple 'set_var'(s)
 
 ;==========================AUTH SECTION OPTIONS=========================
 ;[auth]

include/asterisk/res_pjsip.h

@@ -592,6 +592,8 @@ struct ast_sip_endpoint {
 	unsigned int allowtransfer;
 	/*! Method used when handling redirects */
 	enum ast_sip_session_redirect redirect_method;
+	/*! Variables set on channel creation */
+	struct ast_variable *channel_vars;
 };
 
 /*!

res/res_pjsip.c

@@ -665,6 +665,14 @@
 						set to <literal>sdes</literal> or <literal>dtls</literal>.
 					</para></description>
 				</configOption>
+				<configOption name="set_var">
+					<synopsis>Variable set on a channel involving the endpoint.</synopsis>
+					<description><para>
+					        When a new channel is created using the endpoint set the specified
+						variable(s) on that channel. For multiple channel variables specify
+						multiple 'set_var'(s).
+					</para></description>
+				</configOption>
 			</configObject>
 			<configObject name="auth">
 				<synopsis>Authentication type</synopsis>

res/res_pjsip/pjsip_configuration.c

@@ -761,6 +761,43 @@ static int t38udptl_ec_to_str(const void *obj, const intptr_t *args, char **buf)
 	return 0;
 }
 
+static int set_var_handler(const struct aco_option *opt,
+	struct ast_variable *var, void *obj)
+{
+	struct ast_sip_endpoint *endpoint = obj;
+	struct ast_variable *new_var;
+	char *name = ast_strdupa(var->value), *val = strchr(name, '=');
+
+	if (!val) {
+		return -1;
+	}
+
+	*val++ = '\0';
+	if (!(new_var = ast_variable_new(name, val, ""))) {
+		return -1;
+	}
+
+	new_var->next = endpoint->channel_vars;
+	endpoint->channel_vars = new_var;
+
+	return 0;
+}
+
+static int set_var_to_str(const void *obj, const intptr_t *args, char **buf)
+{
+	struct ast_str *str = ast_str_create(MAX_OBJECT_FIELD);
+	const struct ast_sip_endpoint *endpoint = obj;
+	struct ast_variable *var;
+
+	for (var = endpoint->channel_vars; var; var = var->next) {
+		ast_str_append(&str, 0, "%s=%s,", var->name, var->value);
+	}
+
+	*buf = ast_strdup(ast_str_truncate(str, -1));
+	ast_free(str);
+	return 0;
+}
+
 static void *sip_nat_hook_alloc(const char *name)
 {
 	return ast_sorcery_generic_alloc(sizeof(struct ast_sip_nat_hook), NULL);
@@ -1483,6 +1520,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
 	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "dtls_setup", "", dtls_handler, dtlssetup_to_str, 0, 0);
 	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, 0, 0);
+	ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "set_var", "", set_var_handler, set_var_to_str, 0, 0);
 
 	if (ast_sip_initialize_sorcery_transport(sip_sorcery)) {
 		ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
@@ -1588,6 +1626,7 @@ static void endpoint_destructor(void* obj)
 	endpoint->pickup.named_callgroups = ast_unref_namedgroups(endpoint->pickup.named_callgroups);
 	endpoint->pickup.named_pickupgroups = ast_unref_namedgroups(endpoint->pickup.named_pickupgroups);
 	ao2_cleanup(endpoint->persistent);
+	ast_variables_destroy(endpoint->channel_vars);
 }
 
 static int init_subscription_configuration(struct ast_sip_endpoint_subscription_configuration *subscription)