[asterisk-users] Reinvites and SIP/RTP

[Noah Miller]

Hi Adrian -

> When I use re-invite, does the Asterisk server stay in the SIP conversation,
> and just RTP traffic diverts, or does the SIP transfer away from the A*k
> server too ?

I'm sure somebody will correct me if this is wrong, but I believe the
signalling must stay with asterisk, as asterisk needs to know if it
should provide any services for the call (music on hold, transfer,
etc).


- Noah