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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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