We are trying to turn up an XO ESIP service using FreePBX. This setup is different than I have worked with in the past. In the past we were NATing through a firewall to the trunk provider.
In this case they have a router for their MPLS/ESIP product. The ESIP on their router goes into a NIC on our PBX. I have assigned an IP address to that NIC and can ping the router as well as the trunk IP that they gave me. I have tried the following peer details, I can get a connection to the trunk but I can’t get an outbound call to work and I don’t even know which of the ones below would be correct.
Peer details 1
canreinvite=yes
dtmfmode=rfc2833
host=
outboundproxy=
progressinbound=yes
qualify=300
type=peer
disallow=all
allow=ulaw
Peer details 2
host=
qualify=8000
insecure=very
type=friend
dissallow=all
allow=ulaw
Peer details 3
type=peer
qualify=8000
nat=no
peer=
disallow=all
context=from-trunk
canreinvite=no
allow=ulaw
The user context in all cases has been as follows.
qualify=yes
context=from-trunk
The other thing I am unsure about is the SIP network settings (Settings > Asterisk SIP Settings) we are not doing NAT as far as I can tell because the PBX IP is 10.100.1.25 and the XO router is 10.100.1.254?
I have tried yes, no, and never for the NAT settings. I have added the local networks of 10.100.1.0/24 along with the networks of the phones. DIf I set NAT to no or never it says my call can’t be completed. If I set it to yes I hear nothing.