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Thanks. Also check out OpenSIPS new Re-Homing Tag features. Its really nice

AWS and initially I was binding to public IP. However I ended up switching to a private IP on the cluster and sticking OpenSips in-front of it as the only internet facing servers

I developed this system where you can drag and drop to create anything you can dream of. What would you create? What would you do with this tech? (image below) :)

@DigitalDaz Yes all I'm trying to do is proxy SIP + RTP to a server behind a firewall. Don't want to have to worry about registrations, phone numbers, etc. Just have the SIP/RTP proxy blindly proxy data to one server. What do you recommend is the easiest way to do this if you wouldn't bother...

@DigitalDaz - If I wanted to add RTP proxy to this is it possible in the same "dumb" way where RTP Proxy and Kam doesnt care about users, extensions, phone numbers, etc? I would like to add RTP Proxy in the way where it simply passes the media to Fusion behind a firewall on a private network...

Thanks DigitalDaz. Just to clarify. The Conferencing works when dialing an outside phone number that routes to the PBX. Reason being is the "apply-inbound-acl" maps the trusted IP's to the new context "cluster-in". However dialing the conference extension from the softphone directly doesn't...

Having multiple servers active in a SRV setup seems more scalable and could even do routing based on DNS response times so users get the closest server. Even with a SRV setup weighted so its more Active/Passive it's not guaranteed, meaning any blip in someones internet connection could cause...

In a clustered 2 server fusionpbx setup where both servers share the same database and using DNS for load balancing; how can I make a conference room "sticky" so all callers attempting to enter the conference room are guaranteed to be on the same server to talk to each other. So for example...

Solved this! For anyone else that has this issue here is the solution; Modify dialplan local_extension. Change the bridge line to "sofia_contact ${destination_number}@${domain_name}" This seems get the users registration info with the correct fs_path

Update with more info; In the FS logs I see the correct fs_path where it says "Parsing global variables" However immediately after it says "Setting proxy route to sofia/internal/100@mainsipdomain.com" Then the INVITE is sent to the sip domain not the domain of the node from fs_path

I have a two FS node setup. Both sharing the same DB. fs_path is set to true. I see fs_path set correct when running the command "show registrations" on each node. My issue is when I have extension 100 registered on node1 and extension 200 registered on node2. Extension 100 and 200 can not call...

Use an AWS EFS drive. Mount it on both servers to /var/lib/freeswitch/ and /usr/share/freeswitch/sounds/ This should cover most everything you need in terms of syncing files

Ok thanks phonesimon. I tested this and can confirm that works. I was not expecting it to work when both servers happened to bind to the same IP on the internal profile however it works with calls going to both servers. Now the only thing I can not figure out how to test is the client side...

Ok thanks phonesimon. I'm just stuck on these two parts if you or anyone knows the answer to this; 1) What are you using for external_sip_ip and ext-sip-ip? Should it be the main SRV domain "host:sip.domain.com" or something else? 2) Do I need two internal profiles so each server binds to its...

I'm trying to follow this setup. I have two servers with a different hostname. (server1.domain.com and server2.domain.com) and one SRV/A domain of sip.domain.com). Both servers share the same Freeswitch and Fusion DB. Few questions; 1) What are you using for external_sip_ip and ext-sip-ip...