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/13.6.0 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.