[asterisk-users] How to setup SIP so that RTP traffic flows from Source to destination

[Shaun Wingrin]

The setup is as follows: SIP phone registers via international link to Asterisk Box 1 and calls mean't for termination on Asterisk Box 2 via Zaptel Channels need to be hairpinned from Box 1 to 2. How is sip.conf configured on Box 1 and 2 so that we don't get an error: "Failed to authenticate user" when 1's extensions.conf uses SIP to dial Asterisk Box 2 . How do we ensure that RTP traffic flows from SIP phone registering at 1 directly to 2 without first passing through 2?

Tx

Shaun 
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