[SR-Users] Invite - negotiate to asterisk as peer not kamailio

Pstn->Kamailio->dispatcher->asterisk When calls come in, they are sent to asterisk, and the peer is negotiated as Kamailio. How can I have different peer settings, for example dtmf? I'm thinking I want the call to come in, then when the invite is sent to asterisk, the invite should be from the pstn-peer and not Kamailio, make sense? Then I can have separate peers, with their own dtmf/audio settings, but I see the invite as: INVITE sip:+1866NXXNXXX@<kamailio<sip:+1866NXXNXXX@%3ckamailio> ip>:5060 SIP/2.0 Via: SIP/2.0/UDP <Kamailio ip>;branch=z9hG4bKc087.12db3c91.0 Via: SIP/2.0/UDP <pstn-peer ip>:5060;branch=z9hG4bK0cB824c55100a4f2b8b From: <sip:+1NXXNXXXXXXXX@<pstn-peer ip>:5060;isup-oli=0>;tag=gK0c731df8 To: <sip:+1NXXNXXNXXX@<kamailio ip>:5060> Call-ID: 1024240378_131584110@<pstn-peer ip> Any help or suggestions are very welcome.. Matt Scott