Anybody use freeswitch as sip server of polycom mcu (rmx1500) ? It is ok when call in from endpoint (hdx6000) , but fail when call out from polycom mcu to endpoint . when mcu call out to endpoint, ringing and ack is ok,but after ack ,endpoint invite mcu mcu(10.15.40.58) ---> FS(10.65.6.144) ----> endpoint (10.71.81.15)recv from tcp/[10.71.81.15]:5060 ------------------------------------------------------------------------ INVITE sip:mod_sofia@10.65.6.144:5060;transport=tcp SIP/2.0 From: <sip:1005@10.71.81.15:5060;transport=tcp>;tag=plcm_267848146-3395 To: "ALIPAY_MCU" <sip:test@10.65.6.144>;tag=r9UvSS7DQ51UFsend to tcp/[10.15.40.58]:5060 ------------------------------------------------------------------------INVITE sip:test@10.15.40.58:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.65.6.144;branch=z9hG4bKSjHv7aU76y71H Max-Forwards: 69 From: <sip:1005@10.65.6.144>;tag=Dm347F7QQQcXF To: ALIPAY_MCU <sip:test@alibaba-inc.com:5060>;tag=rmx2k_4122954403-2363 recv from tcp/[10.15.40.58]:5060 ------------------------------------------------------------------------ SIP/2.0 481 Call Leg/Transaction Does Not Existdetail log file： mcu_call.rtf <http://freeswitch-users.2379917.n2.nabble.com/file/n7595199/mcu_call.rtf> Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-out-fail-when-using-Frees... Sent from the freeswitch-users mailing list archive at Nabble.com.