Sip.conf and allowguest behavior


#1

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):

[general]
context=default
allowguest=no
autocreatepeer=no
port=5060
bindaddr=0.0.0.0
srvlookup=yes

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.


#2

Which version of Asterisk are you running?


#3

1.07 on Debian Sarge


#4

I think, but am not 100% sure, that this feature was added as a part of v1.2.


#5

I’ll look into that and maybe build a system with 1.2. I’ll post the results.