I have inbound SIP trunks. All extensions are PJSIP.
In Asterisk SIP Settings - Security Settings, if I do not “Allow Anonymous Inbound SIP Calls” and “Allow SIP Guests”, all inbound calls will not be successfully picked up by the FreePBX.
As soon as I changed it to “Yes”, inbound calls via our SIP Trunks work as expected.
In Call Detail, I can see the “Channel Name” showing as PJSIP/anonymous-0000000nn where nn are numbers. with context = from-sip-external
How to “make” inbound calls from the registered SIP trunks not an anymous?
Obviously, this is a security concern.
You don’t have the trunk setup to accept/match the providers IPs. You need to add all the IPs they could send calls from.
Are your trunks PJSIP or chan_sip?
Trunks are chan_sip.
Now, I configured some PJSIP trunks (basically the same providers) but left them disabled, the issue to have gone away.
Your trunk provider is sending the traffic to you on the port where PJSIP is listening.
My suggestion is to be done with chan_sip altogether and remove it from your configuration, using only PJSIP for both phones and trunks.
It is not. It is just a misconfiguration, and best solved in my opinion by simplifying.
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