Disable authentication on incoming SIP trunk? Where and How?

I am a new user of Asterisk but a very experienced SIP user (Kamailio). My system config is:

OS Version:
Linux xxxxxxxx 4.2.0-19-generic #23-Ubuntu SMP Wed Nov 11 11:39:30 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux
Asterisk Build:
Asterisk GUI-version : SVN--r5219

I have several accounts linked to different DID on my (Asterisk) SIP Trunk provider. I can make outgoing calls using SIP origination (I don’t use IAX)

Incoming calls are rejected with (in the log)

chan_sip.c: Failed to authenticate device <…>

Researching the web, this can be fixed by putting in a line in the correct trunk section of sip.conf.


My problem is that /etc/asterisk/sip.conf doesn’t seem to be used. It seems like much of the config in my present installation is in some database that I can manage with the web-page GUI. I have checked by text searches in /etc/asterisk and none of the trunk names are present. This leads me to believe the config is in a database.

Where and how should I put in the configuration to prevent authentication requirements for incoming calls from my trunk provider IP net?

NB: The physical setup works with Kamailio so the issue is getting Asterisk to work rather than solve connectivity and firewall issues.

Upload your sip trunk options, just with a decription noone can help you.

It is not clear whether you are using Asterisk Realtime Architecture, or a GUI that writes the configuration over AMI. Please identify the GUI (although, if it isn’t FreePBX it is unlikely to be supported anywhere).

insecure=port,invite is wrong and old fashioned.

To inhibit authentication of incoming INVITEs, you only need insecure=invite. However, on modern versions of Asterisk, a better way to do it is to use remotesecret instead of secret.