I have built a pair of Asterisk servers and have setup my extensions file to transfer calls from PBX 1 over to PBX 2. However, I have not added a user setup to the sip.conf file on PBX 2. My understanding is that with “allowguest=no”, that this call should be rejected because it cannot be authenticated. However, it is being allowed to process and get to the voicemail running on PBX 2. My sip.conf on PBX2 is as follows (this is the whole thing, I have not defined any other channels):
Am I missing something here? I am working on a deployment and I don’t want unauthenticated systems able to relay their calls through my PBX.