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

tony niexuping at gmail.com
Thu Sep 26 06:19:41 MSD 2013


    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 



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/call-out-fail-when-using-Freeswitch-as-sip-server-of-polycom-mcu-tp7595199.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130925/2d46431a/attachment.html 


Join us at ClueCon 2013 Aug 6-8, 2013
More information about the FreeSWITCH-users mailing list