Additional Configurations after disabling SRTP

FreePBX Configuration
1

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

Extension settings
transport 0.0.0.0-udp
allow non encrypted audio also set to yes.

Asterisk log just shows one line:
[2023-02-06 18:11:26] ERROR[20471] 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.

2

For the extension, set Media Encryption to None.
Check that the SBC side also is set for no media encryption.

If you still have trouble, at the Asterisk command prompt type
pjsip set logger on
paste the Asterisk log for a failed call at pastebin.freepbx.org and post the link here.

3

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