Nathan Goodwin wrote:
Does anyone have any ideas how I could fix this, this is sort of important, if it's just me because of my NAT causing it, would doing so part forwarding and disable NAT support on asterisk and the Sipura fix this problem?
It's almost impossible to fix this problem. Here's the scenario:
Your SPA-2000 initiates a call to the * server, and then * initiates a call to your provider. When the provider answers, * tells the SPA-2000, and it starts sending RTP to *. By doing so, your NAT/firewall expects to receive packets back from the _same IP address and port they were sent to_. While * is still in the media path, this is how it works, and things are fine.
However, when * tries to re-invite the SIP provider to send audio directly to your SPA-2000, the packets now arrive from a different IP (and probably a different port number). Any decent NAT/firewall will drop them on the floor. Thus, no audio from the provider.
It is _possible_ for this to work if * happens to reinvite your SPA-2000 _first_, and it starts sending audio directly to the SIP provider, thus opening a different IP/port combination through the firewall. However, this is not reliable, and there is no way to force the reinvites to happen in a particular order (or to complete in a particular amount of time).
You can disallow reinvite for your SPA-2000 but leave it turned on for your provider, in which case one direction of the media stream can bypass *. Otherwise, you need a NAT/firewall that understands SIP so it can be aware of the changes as they occur (or you need to use a SIP proxy on the NAT/firewall, like siproxd).
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