Convoluted SIP routing
I have a rather convoluted method of hooking up our SIP PBX (Asterisk) which actually works but I would like to streamline it and possibly add some QoS.
pfSense has the following interfaces:
WAN1 is for general internet/VPN traffic. WAN2 is for Voip traffic.
'BRIDGE' interface is bridged with WAN2.
My Asterisk PBX has two interfaces; one in the LAN and a Public IP in the BRIDGED net. Traffic to the PBX in BRIDGED is filtered to only allow SIP traffic from our SIP Provider. I use a public IP on the PBX to get around the issues with natting SIP.
I would like to introduce some QoS on the PBX but I believe the Traffic Shaper does not support Bridged interfaces.
This leads me to simplifying the PBX routing.
I look forward to any suggestions you can offer so that I can ditch the BRIDGED interface.
Should I use the SIP proxy package if I only have the one PBX comminucating with the outside world?
Any suggested Traffic shaping settings?
Thanks for any suggestions.
You should also post this to the trixbox and pbxinaflash forums. Lots of guru's there.
There really isn't a need to put the asterisk on the internet…It's safer to get it behind the firewall and port forward.
I've successfully established many remote sip extensions with no issues. I have pbx in a flash at the main site, behind a pfsense box. The remote extensions are behind the run of the mill dsl/cable modems. Nothing fancy. Phones have been Snom's and Mitels.
If you're connecting multiple asterisk boxes just use iax2 connections. Only One port to forward.