@lgaetz Thanks for your reply.
I worked with Sangoma SIPStation support yesterday and today and was able to get outbound and incoming calls working with SIPStation by making changes to the NAT rules in pfSense. Support referred me to the following document:
Following instructions here:
https://docs.netgate.com/pfsense/en/latest/nat/configuring-nat-for-voip-phones.html
Setting Static Port using Hybrid Outbound NAT
To disable source port rewriting, the Static Port option must be used on outbound NAT rules. When crafting these rules, be as specific as possible with the source, destination, and destination port to avoid problems with other traffic
- Navigate to Firewall > NAT on the Outbound tab - Done
- Select Hybrid Outbound NA - Done
- Click Save - Done
- Click Add with the up arrow to add a rule to the top of the list - Done
- Set Interface to WAN - Done
- Set the Protocol to match the desired traffic (e.g. UDP) - Done
- Set the Source to match the local source of traffic, such as LAN Net or a specific device such as a game console IP address, or an alias containing multiple such devices - Source: 172.16.0.175 - Done
- Leave the Source Port box empty, which indicates any - udp/ - Done*
- Set the Destination to match the traffic, if known, otherwise leave set to ‘any’ - * - Done
- Set the Destination Port to a specific port or port alias, if it is known, otherwise leave the box blank for any udp/ - Done*
- Set the Translation Address to Interface Address or an appropriate VIP if needed - - Done
- Check Static Port to indicate that traffic matching this rule will retain the original source port - - Done
- Click Save - Done
- Click Apply Changes - Done
- Navigate to Diagnostics > States - Done
- Enter the IP address of the device in the Filter box if a specific source was used in the rule
- Click Filter - Done
- Click Kill - Done
The pfSense firewall was rebooted. When I run the External Connectivity / Check Connectivity Test in the SIPStation module in FreePBX, firewall status again shows FAIL.
An outgoing call to the PSTN (my mobile) works. Audio works in both directions.
An incoming call from my mobile works. Audio in both directions.
I think we can close this issue. However, can someone please explain why the SIPStation External Connectivity / Check Connectivity test shows a false positive. I have spent hours pulling my hair out on on this.
Sangoma SIPStation Support replied to my last question as follows:
Unfortunately regarding the Connectivity test it is not the best. It does need to be either improved or removed because it does sometimes give a result that is not truly indicative of quality of setup.