you could allow anonymous sip calls and then any call will be allowed in, not sure if that is what you are trying to do but if it is, it is set in the General Screen of the GUI interface.
Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx
Thanks Philippe I have tried doing that but it still does not seem to work.
For some reason the key being sent for MD5 authentication by the PBX seems too short and the call is being rejected.
This is for individual call setup and not the initial SIP trunk registration. The SIP trunk is registered fine its just the re-authentication of the inbound call that fails. I allow 6 sessions on this SIP trunk.
If Asterisk is requesting authentication then it may be that you have a sip configuration setup somewhere from the same host requiring authentication. The call will only end up flowing through the anonymous path if no other context can be found that would match with the originating caller’s IP. So check you other sip configurations.
Philippe Lindheimer - FreePBX Project Lead
http//freepbx.org - IRC #freepbx
it’s happening the same to me.
i allow anon calls, and still i get the same 407 error on a specified trunk.
one thing i see is that on the logs there is a line:
To: sip:[email protected]
where that is the contact they receive from me.
the strange thing is that “s”
That’s it! Thanks (I was pulling my hear for two days)! But…
… if I remove username and secret from PEER Details section, then I can’t make a call out, but I can receive calls. Is there any way to define that I don’t want authentication requests only for incoming calls.