Does anyone have documentation on how to configure one FusionPBX (A) to relay calls to PSTN from another FusionPBX(B)? For example, FusionPBX(B) has no gateways, or possibly just a few POTS lines for limited local dialing, and 911. But for 99% of its calls, FusionPBX(B), sends the call invites to FusionPBX(A), which needs to be configured to “trust all calls” from FusionPBX(B) and route all of those PSTN calls to one of its carriers. I am good on all dial plans on each of the two systems. What I can’t seem to get past is how to make system A “trust” system B and process the SIP Invites.
You need not read any further if you have any information you can send me, below I detail my troubles and trifles of how I tried to set it up. The only info I could find on the “docs.fusionpbx.com” site was how to configure an ACL, but I don’t recall the doc indicating for what purposes or configurations would you use ACL’s (sip calls local, sip calls PSTN, http access only?)
What I’ve tried (all of this I tried only on the receiving/relay side, because the PSTN invite always arrived to the relay site, so I never changed anything at the originating site. On the originating site, I set up the relay site as a Gateway, and sent all outbound PSTN calls to that GW, which got the PSTN invite over to the relay site sngrep window looking correctly formatted, so is why I never changed any configuration on the originating site.
So on FusionPBX(A) / relay site only:
Neither worked. Took wire-shark caps on both FusionPBX’s to confirm findings. The SIP invite ALWAYS shows in receiving system’s sngrep screen. But Symptoms ranged from no reply at all in the captures going back to the originating site, to 403 “forbidden” sip message back to originating PBX. Or I would see “Destination Unreachable, port unreachable” ICMP message going from the relay/receiving site back to the originating site. Now the ICMP message hints of Fail2Ban, but confirmed that was NOT the reason, and IP Tables config remains good.
In all cases the invite arrives at the relay site, and always shows in the SNGREP window, but via all config trials above, it’s either FreeSWITCH fs_cli not trusting the inbound call, or my relay site sending ICMP port 83 Destination Unreachable, Port Unreachable (same message as if Banned but it is not banned), or other 4xx messages sent back to the originating system.
The CLOSEST I got – I had calls from originating site into relay site, but being sent to an internal extension, company directory, or local extension. I COULD even get the call out to PSTN, but with a STATIC entry – dial my cell phone, via routed to an internal phone that had call coverage to my cell phone. So I seemed to prove the concept that the call can route from system A to system B to PSTN, just can’t find the recommended configuration to relay PSTN calls between FusionPBXs.
Any guidance , reference links, anything – greatly appreciated! I’ve configured this on all other VoIP PBX’s I’ve supported in the past, but new to FusionPBX and hitting all the snags!
You need not read any further if you have any information you can send me, below I detail my troubles and trifles of how I tried to set it up. The only info I could find on the “docs.fusionpbx.com” site was how to configure an ACL, but I don’t recall the doc indicating for what purposes or configurations would you use ACL’s (sip calls local, sip calls PSTN, http access only?)
What I’ve tried (all of this I tried only on the receiving/relay side, because the PSTN invite always arrived to the relay site, so I never changed anything at the originating site. On the originating site, I set up the relay site as a Gateway, and sent all outbound PSTN calls to that GW, which got the PSTN invite over to the relay site sngrep window looking correctly formatted, so is why I never changed any configuration on the originating site.
So on FusionPBX(A) / relay site only:
- Set iptables to accept all from originating node.
- Created ACL entry on system A with CIDR (x.x.x.x./32) IP of system B. I set that entry up first in Domains ACL only. Later when it didn’t work I set up the entry in Lans ACL only. Later when that didn’t work I set it up in BOTH. Was careful in ACLs NOT to use ‘Domain’ field – always blank, only configured CIDR entries.
Neither worked. Took wire-shark caps on both FusionPBX’s to confirm findings. The SIP invite ALWAYS shows in receiving system’s sngrep screen. But Symptoms ranged from no reply at all in the captures going back to the originating site, to 403 “forbidden” sip message back to originating PBX. Or I would see “Destination Unreachable, port unreachable” ICMP message going from the relay/receiving site back to the originating site. Now the ICMP message hints of Fail2Ban, but confirmed that was NOT the reason, and IP Tables config remains good.
- I abandoned ACL entries (removed all of them), and went a different way: On my relay site (FusionPBX A) I configured a GATEWAY using FusionPBX B’s info. Thus, I was telling my relay site that my originating FusionPBX B is a GATEWAY in which I will be receiving calls. Configured for No Reg, but same way as our config for real inbound Gateway Provider.
In all cases the invite arrives at the relay site, and always shows in the SNGREP window, but via all config trials above, it’s either FreeSWITCH fs_cli not trusting the inbound call, or my relay site sending ICMP port 83 Destination Unreachable, Port Unreachable (same message as if Banned but it is not banned), or other 4xx messages sent back to the originating system.
The CLOSEST I got – I had calls from originating site into relay site, but being sent to an internal extension, company directory, or local extension. I COULD even get the call out to PSTN, but with a STATIC entry – dial my cell phone, via routed to an internal phone that had call coverage to my cell phone. So I seemed to prove the concept that the call can route from system A to system B to PSTN, just can’t find the recommended configuration to relay PSTN calls between FusionPBXs.
Any guidance , reference links, anything – greatly appreciated! I’ve configured this on all other VoIP PBX’s I’ve supported in the past, but new to FusionPBX and hitting all the snags!
- KangarooWho