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

[Michael Jerris]

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 at 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 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 Thanks 
> View this message in context: call out fail when using Freeswitch as sip server of polycom mcu
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