[Freeswitch-users] call out fail when using Freeswitch as sip server of polycom mcu

[tony]

    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 at 10.65.6.144:5060;transport=tcp SIP/2.0  From:
<sip:1005 at 10.71.81.15:5060;transport=tcp>;tag=plcm_267848146-3395   To:
"ALIPAY_MCU" <sip:test at 10.65.6.144>;tag=r9UvSS7DQ51UFsend to
tcp/[10.15.40.58]:5060  
------------------------------------------------------------------------INVITE
sip:test at 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 at 10.65.6.144>;tag=Dm347F7QQQcXF   To: ALIPAY_MCU
<sip:test at 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 



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