[asterisk-dev] Asterisk and multiple SIP registrationstothesame host

[Nick Lewis]

>We always have the ACLs. Surely you use them. ;-)
Um yes well er... I am of course only concerned with others who are not
power users such as I!? Perhaps a call could be treated as a guest call
if it did not match one of the ITSP's ip addresses in the peer
definition (including "outbound" proxy when itsp uses a nat proxy)

>That's my peerfailover stuff really. I have it for 1.4 but haven't made
it work for trunk after the inclusion of the TCP stuff that broke many
things in my code.
I am not familiar with the peerfailover stuff but by nat proxy I mean an
ITSP device other than the sip server that provides extra functionality
specifically to overcome the customer's nat firewall. The nat proxy acts
as both the outbound proxy and the inbound proxy for the service. 

>That's not an option, that's the only way to find out where the call
goes to. Otherwise we're stuck with "s" in the context since we have no
extension.
>Or we could have a "defaultexten" in the context to go to. Hmm.
Any of these approaches will be more understood by the users than just
leaving it to the dialplan to look at the to-header. The dialplan should
not have to understand the details of the channel technology.

>You mean the register a default domain route stuff? That's still
undergoing a lot of changes, we need to see where it's heading first.
I think I agree with you - I am still a bit uncomfortable with it and
there are no ITSPs using this approach. Apparently it gets over a
problem when a call is diverted in the network and the to-header
contains the original destination rather than the DID. In practice ITSPs
just ditch the original destination for a diverted call and replace it
with the DID. The harm done is minor.

N_L

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