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 fileļ¼
<a href="http://freeswitch-users.2379917.n2.nabble.com/file/n7595199/mcu_call.rtf" target="_top" rel="nofollow" link="external">mcu_call.rtf</a>
Thanks
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