By default in FreePBX, pjsip listens on port 5060 and chan_sip is on port 5160. On the FXO tab, Primary SIP Server should be
Outbound Proxy should be blank.
(If you have changed Port to Listen On for pjsip and/or Bind Port for chan_sip, please provide details.)
The Neufbox plays a non-standard dial tone that the HT may not recognize, so set Wait for Dial-Tone to No. Also set Stage Method to 1.
You have the HT registering to the PBX but also have a static configuration for the trunk. I don’t know whether that’s a conflict. I recommend using registration; registered status lets you know that basic communication between the devices is working. Please try this config:
Also, Trunk Name on the Outgoing tab should be 03XXXXXXXX (same as username).
Reboot the HT, wait for it to come ready, its Status page should show the FXO as Registered and Idle.
If not, we will troubleshoot that first. What, if anything appears in the Asterisk log when it attempts to register?
If registration is successful, try an outgoing call. If that fails, at the Asterisk command prompt, type
sip set debug on
retry the call, paste the relevant section of the Asterisk log at pastebin.freepbx.org and post the link here. Also, report what you hear on the call attempt.
If registration is successful, try an incoming call. If that fails, post the log as above and report what the caller hears.