ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app

This patch adds a new feature to ARI to redirect a channel to another server,
and fixes a few bugs in PJSIP's handling of the Transfer dialplan
application/ARI redirect capability.

*New Feature*
A new operation has been added to the ARI channels resource, redirect. With
this, a channel in a Stasis application can be redirected to another endpoint
of the same underlying channel technology.

*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP
stack:
(1) The existing .transfer channel callback had the limitation that it could
    only transfer channels to a SIP URI, i.e., you had to pass
    'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
    still supported, it is somewhat unintuitive - particularly in a world full
    of endpoints. As such, we now also support specifying the PJSIP endpoint to
    transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
    updating its Contact header. Alas, that resulted in the forwarding
    destination set by the dialplan application/ARI resource/whatever being
    rewritten with very incorrect information. Hence, we now don't bother
    updating an outgoing response if it is a 302. Since this took a looong time
    to find, some additional debug statements have been added to those modules
    that update the Contact headers.

Review: https://reviewboard.asterisk.org/r/4316/

ASTERISK-24015 #close
Reported by: Private Name

ASTERISK-24703 #close
Reported by: Matt Jordan



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 20 insertions, 2 deletions

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index cd8e39623..5476fa66c 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -1329,6 +1329,8 @@ static void transfer_redirect(struct ast_sip_session *session, const char *targe
 	pj_str_t tmp;
 
 	if (pjsip_inv_end_session(session->inv_session, 302, NULL, &packet) != PJ_SUCCESS) {
+		ast_log(LOG_WARNING, "Failed to redirect PJSIP session for channel %s\n",
+			ast_channel_name(session->channel));
 		message = AST_TRANSFER_FAILED;
 		ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
 
@@ -1341,6 +1343,8 @@ static void transfer_redirect(struct ast_sip_session *session, const char *targe
 
 	pj_strdup2_with_null(packet->pool, &tmp, target);
 	if (!(contact->uri = pjsip_parse_uri(packet->pool, tmp.ptr, tmp.slen, PJSIP_PARSE_URI_AS_NAMEADDR))) {
+		ast_log(LOG_WARNING, "Failed to parse destination URI '%s' for channel %s\n",
+			target, ast_channel_name(session->channel));
 		message = AST_TRANSFER_FAILED;
 		ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
 		pjsip_tx_data_dec_ref(packet);
@@ -1382,14 +1386,28 @@ static void transfer_refer(struct ast_sip_session *session, const char *target)
 static int transfer(void *data)
 {
 	struct transfer_data *trnf_data = data;
+	struct ast_sip_endpoint *endpoint = NULL;
+	struct ast_sip_contact *contact = NULL;
+	const char *target = trnf_data->target;
+
+	/* See if we have an endpoint; if so, use its contact */
+	endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", target);
+	if (endpoint) {
+		contact = ast_sip_location_retrieve_contact_from_aor_list(endpoint->aors);
+		if (contact && !ast_strlen_zero(contact->uri)) {
+			target = contact->uri;
+		}
+	}
 
 	if (ast_channel_state(trnf_data->session->channel) == AST_STATE_RING) {
-		transfer_redirect(trnf_data->session, trnf_data->target);
+		transfer_redirect(trnf_data->session, target);
 	} else {
-		transfer_refer(trnf_data->session, trnf_data->target);
+		transfer_refer(trnf_data->session, target);
 	}
 
 	ao2_ref(trnf_data, -1);
+	ao2_cleanup(endpoint);
+	ao2_cleanup(contact);
 	return 0;
 }