I've searched this forum and found this brought up a few times before, but I'm still not sure I understand it completely.
I've been adding some failover stuff to my asterisk setup since the Callcentric outage, and now that I'm into it and testing things, I'm noticing some problems.
With allowguest=no in sip.conf, if I try calling my 1777 number from a softphone, I get an error like this in asterisk:
handle_request_invite: Sending fake auth rejection for device "Soft Phone" <sip:17778888888@callcentric.com>;tag=123456789
Then if I set allowguest=yes in sip.conf and try again, I get an error like this instead:
handle_request_invite: Call from '' (204.11.192.171:5060) to extension '17779999999' rejected because extension not found in context 'default'.
It's clear from the second example that the call is coming from a different IP address and Port than what I'm registered to.
My asterisk server is behind NAT with no port forwarding, so how does the call even come through from a different IP address and Port than what I'm registered to?
That's the first thing I'm wondering about.
I'm also pretty sure that's the cause of the error, too (incoming calls from servers that I'm not registered to).
It seems I can get it to work by adding a peer for each of the 20 Callcentric servers with entries in sip.conf like:
[callcentric1]
type=peer
host=alpha1.callcentric.com
context=callcentric-trunk
But that seems fragile since it will break if callcentric adds more servers or changes the host names.
What do other people do to get it to work?
Also, I should note, I'm only testing it by calling my 1777 number and iNum right now, since my Callcentric DID is still down. Maybe incoming calls to the real DID all come in from the server you're registered to?