asterisk users - Sep 2008 - How to setup SIP so that RTP traffic flows from Source to destination

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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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/

> 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?
I think if you set up a peer for Box 1 on Box 2, and set insecure=port  
on those peers, that it will not try to auth calls that are from your  
other asterisk box.  Of course, you'd have to make sure in your  
diaplan that you restricted access to those calls appropriately.  For  
the RTP, setting canreinvite=yes one peers that you want to be able to  
send media directly to each other should allow the RTP behavior you  
are looking for, but keep in mind that if there are any NATs between  
the phones, things can get messy in a hurry.