[Asterisk-Users] SIP CANREINVITE

[Martin Sutherland]

I have a number of internal SIP phones and a number of external SIP clients with the server running Asterisk on the boundary between the two. ie the server has two network cards with an internal private address and an external public address. For security reasons no routing is allowed between the two thus no internal phone can talk directly to external phone or visa versa. I am very happy with this arrangement but would like to reduce the load on Asterisk where possible by  allowing reinvites. Obviously only internal phones can talk with each other and external phones can talk directly with other external phones. The problem is the logic in Asterisk doesn't allow this.

Ideally Asterisk would look at both sides of a call that is has setup and realise that both were internal and thus could be potentially reinvited or that both are external and could thus be reinvited. Where the source or destination was not on the same network then reinvites would not be possible.

Of course everyone's network is different and this may not work for some, but wouldn't this be a good idea?