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

[Anthony Francis]

Shaun Wingrin wrote:
> 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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This happens through a sip re-invite, the problem you seem to be having 
is that box 1 is not authenticated to send calls to box 2.

Anthony

/"Everything should be as simple as possible, but no simpler" - Albert 
Einstien/