<html><head><meta http-equiv="Content-Type" content="text/html charset=us-ascii"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi<div><br></div><div>My application is for mobile (MVNO) users making calls, which will generally end up on the PSTN via our carriers.</div><div><br></div><div><font face="Courier New"> MVNO Carrier --> Our Edge Switch --> Our PSTN Switch --> Our carrier's switch</font></div><div><font face="Courier New"> |</font></div><div><span style="font-family: 'Courier New'; "> |---> Our internal routing switch</span></div><div><br></div><div>The issue is with our PSTN switch and the fact that it is not staying in the SIP signalling path, so when the call ultimately between our MVNO carrier and outbound Carrier is established (200 OK) the MVNO carrier and PSTN carrier begin talking to each other.</div><div><br></div><div>When the MVNO carrier issues a BYE to the outbound carrier, the outbound carrier does not then receive this packet as they are firewalled (and always will be).</div><div><br></div><div>What is the correct method of relaying calls through Kamailio but not passing on the Contact: header info? I have read that forcing a change of Contact is not the right way.</div><div><br></div><div>Leo</div></body></html>