Hi, I’ve followed the steps in the tutorial named “FreeSBC:Remote_Workers:Configuration_A” on the wiki and I seem to have it working where registrations and signaling is correctly forwarded to an Asterisk server.
In Asterisk using pjsip, it’s easy to have multiple transports and listen on port 5060 using UDP and TCP and 5061 using TLS.
I’m having trouble configuring FreeSBC to do this.
I created several SIP transports, one for each protocol. Do I need to create a corresponding NAP and route for each?
Fixes: #23213 - DHCP MTU is hardcoded at 1500 #23282 - Single port interface should not be readonly #23471 - Vmxnet3 hardware IP checksum not calculated for RTP packets #23600 - Add call-id in the routing script #23602 - FQDN SRV record support
I think I got it working using two NAPs on the wan side just for UDP and TCP for now.
But I’m getting multiple registrations using TCP using the softphone Linphone. It registers fine at first, but every minute or so it tries again. These contacts pile up on FreeSBC and Asterisk until it hits the max contacts allowed on Asterisk.