[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=r9UvSS7DQ51UF

send 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 Exist

detail log file：
mcu_call.rtf
<http://freeswitch-users.2379917.n2.nabble.com/file/n7595224/mcu_call.rtf>  


Thanks 



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-out-fail-when-using-Freeswitch-as-sip-server-of-polycom-mcu-tp7595224.html
Sent from the freeswitch-users mailing list archive at Nabble.com.