I have had a working system for five years, a recent update has changed how freepbx/asterisk is working with VPN connections
I have:
Office-A --> openvpn --> Office-B
10.1.1.15 ext321 10.0.99.2 <->10.0.99.1 192.168.100.15 FreePBX
I have been fighting with one way audio for awhile now. randomly the client will connect with
res_pjsip_registrar.c: Added contact ‘sip:[email protected]:49057;transport=UDP;rinstance=7a57149bf842a1f2’ to AOR ‘321’ with expiration of 60 seconds | |
---|---|
17590 | [2021-11-08 11:32:55] VERBOSE[4557] res_pjsip/pjsip_configuration.c: Endpoint 321 is now Reachable |
And everything works perfectly, audio both ways and no drops
Most of the time though it will connect with
es_pjsip_registrar.c: Added contact ‘sip:[email protected]:49057;transport=UDP;rinstance=89f162414fd14318’ to AOR ‘321’ with expiration of 60 seconds | |
---|---|
17596 | [2021-11-08 11:32:55] VERBOSE[2346] res_pjsip/pjsip_configuration.c: Endpoint 321 is now Reachable |
and obviously sending to the gateway will not allow return audio from the server. At this point I have tried all the various rport, rewrite, etc settings I can find in the forums. Nothing seems to change the random behavior