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SIP proxy info?

I am hoping someone can provide useful information as to how this is supposed to be used.  What I have: an asterisk (pbx in a flash) server behind the ASG.  UDP/5060 and UDP/10000:20000 forwarded to the pbx with a DNAT rule.  I have a QoS rule giving precedence to traffic marked as low delay from the pbx.  I haven't seen any posts recently that really talk much about the SIP helper, although I get the impression from a post of Gert's from a couple of years ago that if I used this, almost all of the config stuff would not be needed.  Can anyone clarify this?


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