Hi,everyone For voice communication I use a SIP+RTP proxy together with SER. For PSTN calls SER routes the INVITE messages to the RTP proxy. Everything works fine there, but when the call is ended by one client (which is connected to SER) or the PSTN user, the BYE message arrive to SER but dont forward it. If our RTP proxy does not see the message coming from SER, call doesn't end. That's why I need all of the BYE messages to go through SER. How can I do that ? In the documents they said that the bye message loose route if there are ";lr" and If the message should be loose routed then SER should simply relay the message to the next destination as specified in the top-most Record-Route header field .but i have a try and nothing happened! that is my config block: # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" # ------------------------- request routing logic ------------------- # main routing logic route { record_route(); if(message=="BYE") loose_route(); }; my ser is 1.2.3.4:5060 this is my sip message: BYE sip:0000123@19.68.150.20:31562 SIP/2.0^M Via: SIP/2.0/UDP 23.43.53.4:5060;branch=z9hG4bK5a69ae91;rport^M Record-Route: <sip:1.2.3.4:5060;ftag=28118cbe-30554463;lr>^M Record-Route: <sip:12.45.32.61:5060;r2=on;lr;ftag=28118cbe-30554463>^M From: <sip:123456@1.2.3.4:5060;user=phone>;tag=as7232fbd6^M To: "beurself212" <sip:0000123@19.68.150.20:31562>;tag=28118cbe-30554463^M Call-ID: 76ad6e55-86f19c97-MzA1NTQ0NjM(a)192.168.50.50^M Thanks, Jimmy 2008-09-12 邱磊