Fix a random one way audio issue in PJSIP.
[asterisk/asterisk.git] / res / res_pjsip_session.c
index 965f66b..9cb85dc 100644 (file)
@@ -472,6 +472,7 @@ static int handle_negotiated_sdp(struct ast_sip_session *session, const pjmedia_
        successful = ao2_callback(session->media, OBJ_MULTIPLE, handle_negotiated_sdp_session_media, &callback_data);
        if (successful && ao2_container_count(successful->c) == ao2_container_count(session->media)) {
                /* Nothing experienced a catastrophic failure */
+               ast_queue_frame(session->channel, &ast_null_frame);
                return 0;
        }
        return -1;