Hello All,
I am working on a first/test deployment of 4.2.1 and am struggling to make decisions about the network topology. 1 - This system will replace my trusty FreePBX system in my office and will connect to my ITSP (Teliax) via B2BUA. 2 - I want to test and understand the concepts of running a small office phone system with both remote workers and remote branches/offices 3 - I have installed 4.2.1 from the ISO. 4 - I am deploying a pfSense firewall as well 5 - I have a statis IP available for sipX if needed and have all external DNS SRV records correctly configured 5 - I want to understand sipXECS in the context of deploying in small offices with limited disruption to the existing network infrastructure (or, with out the need to completely re-work the infrastructure of an existing office) With all that in mind, my current thinking is to deploy on a separate LAN (VLAN) and letting the sipX server handle DNS/DHCP/NTP for the phones. Questions: 1 - Should sipX have a public IP address behind a 1:1 NAT in pfSense? Or, should it have a public IP directly on the Internet. Or, is it sufficient to have a private IP on the phone VLAN and port forward as needed 2 - If sipX has a private internal IP/subnet - I'm assuming I would run a "Split DNS" structure, with the internal DNS server running on the sipX server pointing to the private IP and my outside records pointing to my public IP 2 - Is it correct thinking to assume that the ultimate goal with this deployment is to allow media streams from point-to-point in as many situations as possible. Is this the correct framework to be thinking about how to lay out the network? 3 - In light of #2 above, using VPNs to tunnel remote workers is not a good solution - correct? Usage Scenario - If I place a Polycom phone at my house (basic DSL connection behind a generic NAT router) - and have it configured with a line registered to my sipX server - if I place an outside call, can I expect that media paths will be established directly to the ITSP? What about an incoming call - caller calls in and receives an Auto-Attendant (RTP is flowing between ITSP and sipX), caller selects an option to call my remote extension at the house (signalling is between sipX and remote Polycom - when call is answered, RTP is between Polycom and ITSP) - callee at the house then transfers the call to an extension at the office (signalling is between the Polycom and sipX/Tranferee Extension, when call is answered, RTP is between ITSP and Transferee Extension). Is this what the "goal" would be - if I have correctly configured my topology and devices? Thanks so much for any help, Dave Redmore Spigot Networks, Inc.
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