[Asterisk-Users] SIP reinvite asterisk and NAT

[Mark Phillips]

If these phones are all to be in a single location I'd deploy a remote 
Asterisk box and run an IAX trunk between remote and local sites. 
That'll save more bandwidth than having a potential 5 individual SIP 
sessions running over your link.

Also, with the addition of an analogue card such as the TDM400 series 
you'll have survivability should your link go down.

If you don't add a phone line to the remote site how will they be able 
to call 911 etc?

Mark

Damon Estep wrote:
> I would like to setup up a remote office with a half dozen or so SIP 
> phones connected to an asterisk server via a WAN link. To conserve 
> bandwidth I would like the phones to be able to re-invite when they call 
> each other.
> 
>  
> 
> The phones will be Polycom, Cisco, or Snom.
> 
>  
> 
> I may or may not use NAT. Seems like the NAT would really mess up 
> re-invites, any experience with that?
> 
>  
> 
> Assuming no NAT, what should be expected in this setup?
> 
>  
> 
> I know the transfer option in asterisk would not work, but I do not 
> think that is a big deal since any re-invited calls would be user to 
> user, with little or no need to transfer.
> 
>  
> 
> As long as the SIP termination peers I am using are set to 
> canreinvite=no then a call between the users and a remote party would 
> not be re-invited, since the peer terminating the call is set to no, 
> correct?
> 
>  
> 
> Can someone share some experiences wit this type of setup? Are there 
> other real issues to look out for or be aware of?
> 
>  
> 
> I am really just trying to avoid having another asterisk box in the 
> remote site to maintain, but do not want to waste bandwidth on calls 
> going across the office.
> 
>  
> 
> Thanks for taking the time to share your wisdom.
> 
>  
> 
>  
> 
>  
> 
> 
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-- 

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com