On Nov 27, 2008, at 3:26 AM, Tony Graziano wrote:
I tried OpenSBC a long time ago and went with Ingate for siptrunks.
I think there is the beginning of a wiki page started for OpenSBC:
http://sipx-wiki.calivia.com/index.php/SipXecs-OpenSBC
The issue you have is NAT traversal. You are trying to define a
siptrunk in your dial plan byy the providers IP address. IF you have
a public IP address on your phones and your sip system with no
firewalls, this would actually work. Because you are on a private
network though, you can't traverse NAT natively with sipx, until the
next major relase ships.
So you need to:
1. Use an ATA to connect to a provider (Vonage, whatever) and
connect the phone line to a FXO gateway that works with sipx
2. Use a SBC or SIP aware firewall to connect to the sip trunk, and
define that device (not the provider directly) and use it as a
gateway within sipx.
3. Use a standard POTS line with an FXO gateway.
SIP doesn't traverse NAT through a firewall easily. Just forwarding
port 5060 won't help. The media uses high ports and is random, plus
a picture is worth a thousand words.
http://freshmeat.net/articles/view/2079/
Something like FreeSWITCH should work as well, right? Aim sipX at
FreeSWITCH and vice versa? I had calls coming and going with
FreeSWITCH... Just thinking out loud here
Chad
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