firstly a big hello to the community since this is my first post and thank you.
I have been using asterisk in combination with freepbx and without for a long time without problems.
Until some days ago, i had my asterisk 13.9.1 compiled pbx with freepbx , working with tls/srtp with my voip clients (mostly softphones on windows and android and some Yealink T26P phones).
Some days ago i upgraded to asterisk 13.11.2 and a strange problem happened.
-I would like prior to everything to inform that my asterisk pbx is behind pfsense configured correctly (have been working for more than a year without issues at all), both for local and outsite voip clients. Also i have read every forum post related to the sip retransmission problems.-
After the upgrade to 13.11.2 i get sip retransmission timeout errors Only in incoming calls and Only when using tcp or tls, both from local and remote extensions. With udp there is no problem at all. With tcp or tls i can make outbound calls without problems but when trying to make a call to an extension or a trunk voip number that rings an extension i get the retransmission timeout and the call hangs.
I have tried a lot of settings both in pfsense and the asterisk server for NAT related issues but without result. Also if that was the issue it would not work with my previous configuration where i had no problems.
I have made a sip debug call from an extension on the outside, calling it’s self, that should ring and put on hold listening the music, like a received call, instead i get the call hang with the error. (with udp it works okay.) Here i posted the debug text https://justpaste.it/yk95 (more than 500 lines, i have edited the IPs’ middle numbers with XXX for privacy).
I have downgraded to astersk 13.9.1 for now and hope that with your help i can resolve this issue and see if that’s an asterisk bug or in the new version there is something else i need to configure.
Sorry for my long message i wanted to be as clear as possible.
Thank you in advance for any help.