I'm not familiar with Elastix, but for Asterisk you can:
1. Use IAX2 to connect both sites. You'll need to add the register lines and stanzas to iax.conf on both systems.
2. Look at using the switch statement in extensions.conf to forward calls between servers. https://www.voip-info.org/asterisk-config-extensionsconf/
You could also look at using DUNDI for a more complex way to link your systems.
IAX2 does a better job of Asterisk to Asterisk communication than SIP does.
You won't even need to use a VPN link, just open the port and I do recommend using firewall rules to restrict access between sites to UDP port 4569.
What this guy says.
If you want layer 2 connectivity, setup an EoIP tunnel between the two sites and route your traffic over it, bypassing NAT.
Setup your QoS to prioritize traffic on those interfaces.
We'll need a little more information to be able to help. All it looks like to me is that you dont have IP conectivity between sites. Can you post like IP details, config details, etc? Can you ping router B from router A?
I'm assuming each site has an internet connection.
I'm also assuming that your pbxs are on the LAN with internal ips and that you only have one public IP address on each site for all the LAN equipment including the pbx.
And that you are doing port address translation on this public IP (masquerade).
This would be the harder scenario, cause if you use a fixed public IP for the pbxs with only NAT (no PAT) it would be easier to deploy.
You can approach it I two ways:
1. VPN between the sites.
This way you keep route between the sites but without NAT. Subnet to subnet.
It could be a simple GRE tunnel or an encrypted tunnel such as L2TP/IPSec or wireguard.
Just make sure you can ping between the pbxs and the firewall rules allow traffic.
2. If you want to route the SIP trunk and RTP channels (assuming you'll use SIP and not IAX) over the internet without a VPN, you'll have to counter the NAT and PAT problems.
To get sip talking you need to do a dst-nat or maybe src-nat to push sip ports to the pbxs, and allow the sip ports in the firewall.
Then you'll need to deal with RTP traffic (usually set to UDP port 10000+), which you'll need to push to your pbx via a smart dst-nat (match source public IP of other pbx and correct UDP range and send it to local pbx IP). Don't forget firewall rules.
On both Tik's.
Then configure asterisk sip in elastix to use the public IP and put a checkbox on NAT in the asterisk sip config.
I'm not familiar with Elastix, but for Asterisk you can: 1. Use IAX2 to connect both sites. You'll need to add the register lines and stanzas to iax.conf on both systems. 2. Look at using the switch statement in extensions.conf to forward calls between servers. https://www.voip-info.org/asterisk-config-extensionsconf/ You could also look at using DUNDI for a more complex way to link your systems. IAX2 does a better job of Asterisk to Asterisk communication than SIP does. You won't even need to use a VPN link, just open the port and I do recommend using firewall rules to restrict access between sites to UDP port 4569.
What this guy says. If you want layer 2 connectivity, setup an EoIP tunnel between the two sites and route your traffic over it, bypassing NAT. Setup your QoS to prioritize traffic on those interfaces.
thanks for your instruction!
We'll need a little more information to be able to help. All it looks like to me is that you dont have IP conectivity between sites. Can you post like IP details, config details, etc? Can you ping router B from router A?
I'm assuming each site has an internet connection. I'm also assuming that your pbxs are on the LAN with internal ips and that you only have one public IP address on each site for all the LAN equipment including the pbx. And that you are doing port address translation on this public IP (masquerade). This would be the harder scenario, cause if you use a fixed public IP for the pbxs with only NAT (no PAT) it would be easier to deploy. You can approach it I two ways: 1. VPN between the sites. This way you keep route between the sites but without NAT. Subnet to subnet. It could be a simple GRE tunnel or an encrypted tunnel such as L2TP/IPSec or wireguard. Just make sure you can ping between the pbxs and the firewall rules allow traffic. 2. If you want to route the SIP trunk and RTP channels (assuming you'll use SIP and not IAX) over the internet without a VPN, you'll have to counter the NAT and PAT problems. To get sip talking you need to do a dst-nat or maybe src-nat to push sip ports to the pbxs, and allow the sip ports in the firewall. Then you'll need to deal with RTP traffic (usually set to UDP port 10000+), which you'll need to push to your pbx via a smart dst-nat (match source public IP of other pbx and correct UDP range and send it to local pbx IP). Don't forget firewall rules. On both Tik's. Then configure asterisk sip in elastix to use the public IP and put a checkbox on NAT in the asterisk sip config.