Connecting FreePBX & asterisk
*Tested on asterisk 13+ with ChanSIP.
Input your own values from your trunks configuration details located in the PBX Connector app via the dashboard. If you need help setting that up, please start here.
Trunk Name: nurango
[nurango] ; add line to asterisk when using command line only, FreePBX uses the "Trunk Name" input box.
type=peerqualify=3400 ; recommended to avoid closure from firewalls "Keep-Alive"
context=from-pstn ; default (try "from-trunk" if doesn't work for you)
srvlookup=yes ; optional if you are using DNS instead of IPs for our Proxy servers
nat=comedia ; Add ",force_rport" if not working. Change to "no" if asterisk isn't behind NAT (external IP)
sendrpid=yes ; to send custom caller-ID in header
The “Incoming” Context is not needed since we have set type=peer
This should be sufficient for 99% of customers using asterisk 1.6 and up. It should be noted that when you use a Peer you do not need to separate the Outbound and Inbound Contexts.
Inbound Call Notes
Since we send inbound calls from multiple IP’s, you will need to “Allow Anonymous Inbound SIP Calls” under “Asterisk SIP Settings” and “Allow SIP Guests” under “Chan SIP Settings”. Create a “Catch All” and send it to “Congestion” so as to terminate unwanted callers not calling to a valid DID.
*Note – Add DIDs as a 10 digit number when configuring Inbound Routes.
For more information on securing your box with a catch-all DID, see our blog article here.
You may also need to Whitelist our IP’s in your firewall. Please contact support for a list of the IP’s if you do not already have them.