Could you post the full sip trace of the failed call, not just one packet.  Also, however you pasted this into the message, it lost all its line breaks so its not readable.

Mike

On Sep 25, 2013, at 10:19 PM, tony <niexuping@gmail.com> wrote:

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=r9UvSS7DQ51UF send 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 Exist detail log fileļ¼š mcu_call.rtf Thanks
View this message in context: call out fail when using Freeswitch as sip server of polycom mcu
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