best way to setup direct calling?
 

i am using various sip accounts & asterisk, however i would like to setup some atas for others to call each other using enum and sipbroker without having to go through a server.

what is the bestway to set this up?

thank you

this would be to save bandwidth on the server end

i figure if you had a static ip it wouldnt be hard, but what about if the connection is dynamic? do any atas support dynamic dns service?

what about our asterisk server, what is the best way to call another office connection without using bandwidth (having the server process) the entire call?
thanks!

Take a look at the SIP Broker Wiki: Inbound Calls Directly to your LinkSys or Sipura. You could expand this further and use ENUM to point directly at your adapter.

thanks for the link

I was going to do a bit googling when I got back to a pc (on axim now)

pdx currently supports direct sip inbound and I can call the extensions directly however I never determined if the calls go through the server.
km

VoXBasic?
 

You could map their PSTN number at e164.org to

username@ipaddress:listening_port

(or in some cases, using dynamic DNS, etc).

as explained here:

voipstuff.net.au » IPDialling


However, considering the hassle of managing changes to IP addresses, setting up dynamic DNS, and potential problems with DNS SRV records (the lack of them), I think it would be easier for you to just set up a VoXBasic account for each ATA.

The basic service does all that you need for what you've described (and more). There's no extra latency, because VoXaLot's RTP proxy is not used for VoXaLot-VoXaLot calls.

If you want to configure something extra afterwards, you can configure their VoXaLot dial plans remotely via the VoXaLot web site.