[dboyce]

Yes, there is nothing that you need to do to support SIP, there is a 'tick' / 'check' box, but even this is ticked for you!

QoS is available on the non ADSL versions (as sold by Voxilla), and will be coming on the ADSL versions (hopefully soon), this QoS works really well.

David

[jrj]

I'm unclear on which functionality distinguishes the normal IX66 from the SIP switch version. $500 seems like a lot for a home user to pay for a little more functionality on a $200 box, especially when it seems that some of that used to be included in the base functionality.

It would be helpful to at least understand exactly what comes at the base level, and what gets added specifically by the SIP switch.

[PhoneBoy]

The base functionality of the IX66 is that it understands SIP, so it will rewrite the SIP packets from internal machines in such a way that they will work through NAT.

The SIP Switch functionality lets you have the IX66 register with other service providers on your behalf and then you register to the IX66 directly. You can then establish a dial plan that allows you to select which provider is used in which circumstances.

[mayoor]

let me understand now it only does manipulation on traffic from inside and not outside .... meaning if a sip calls comes from outside it will not connect to inside appropriately.

You have to have a switch to do that.

I was hoping you can use it simply as a proxy and registrar so inside sip and outside sip can find each other and talk. With a license for 5 users you could have upto 5 (total inside and out) to be able to talk to each other.

I am trying to use ix66 ---NOT SWITCH VERSION so let asterisk do all the routing between pstn/voip trunk and now can have 4 remote extension that can connect to asterisk from IX66 --- which can work as a proxy and registrat

I AM VERY NEW TO VOIP/ASTERISK/LINUX ----just want to know if this scenario will work?