Kamilio 5.5 or opensips 3.2 TCP to UDP Proxy

I have a linux server running asterisk running Chan_sip which only support UDP, and can not be upgraded. The asterisk Server communicates to Phones and SIP TRUNK Providers on UDP port 5060 using the LAN Public IP X.X.X.X interface. I need to have some new phones connect to the Server over TCP Port 5062 I need to have some new phones connect to the Server over TLS Port 5089 We will need to have kamalio or Opensips centos 7 to run on the same server as asterisk, and should allow the following: 1) The Proxy should communicate with the asterisk server on [login to view URL] UDP SIP port 5060. 2) Listen for incoming SIP TCP traffic on 5062 on the LAN IP X.X.X.X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the [login to view URL] interface. 3) ALL TCP SIP traffic on 5062 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC) 4) Listen for incoming SIP TLS traffic on 5089 on the LAN IP X.X.X.X. and proxy this SIP TLS Traffic to and from the Asterisk server on UDP Port 5060 using the [login to view URL] interface. 5) ALL TCP SIP traffic on 5089 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC) 6) The other phones and the sip trunk providers still use UDP 5060 and should not be proxied via Kamailio , and instead should communicate with the Asterisk server directly. Deliverables will provide the complete [login to view URL] or [login to view URL] to allow the remote phone to register on the asterisk server using either TCP or TLS. These phones should be able to make and receive calls just like the current phones connected on UDP Port 5060. instructions on how to create any TLS Certificates needed by Kamailio or OpenSips, will need to be provided. (aka steps for letsencrypt etc) NOTE: the Servers is connected to a public IP address directly, and the phones would be connecting from various remote sites, some of which are behind a nat firewall. The asterisk server has the sip device set to nat=yes. this is working just fine under UDP port 5060. We will need this to continue to work in this environment with the exception that some of the phones may connect (through the proxy) using TCP or TLS. Candidate should have complete understanding of the Proxy solution being provided as well as Asterisk. this project should not involve any additions to the core asterisk system or any asterisk addons. The solution should be a simple SIP Proxy.