I tried to set up TLS and SRTP with a phone that I was playing around with and couldn’t get it to work. No Biggy. But after I went through and disabled all of the things I enabled for TLS/SRTP I am getting a 488 not acceptable here.
Asterisk SIP settings
- Cert Manager Default
- ssl method default
- both verifys are no
Transport 0.0.0.0-udp is the only one turned on.
listening on port 5060
allow non encrypted audio also set to yes.
Asterisk log just shows one line:
[2023-02-06 18:11:26] ERROR res_pjsip_session.c: SBC: Couldn’t negotiate stream 0:audio-0:audio:sendrecv (nothing)
I am routing through an SBC, running a trace on the SBC I see ULAW/ALAW/G729 as offered Codecs which are all enabled on the FPBX side.
I can call from the affected extension registered to FPBX to the other on the SBC just fine. I just can’t call the other way around.
Its almost like there is something that got changed automatically when enabling TLS/SRTP that didn’t get automatically disabled when turning it back off.