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On mediation Server why does not change FROM-sip-bjtest@ch.com.cn to line uri sip-8601012345678@ch.com.cn RRS feed

  • Question

  • In OCS 2007 R2, I config the user bjtest, enable Enterprise Voice, add line uri: +8601012345678,
    in my memory of ocs 2007,when the signaling is sent by mediation server,
    the signaling looks like blew,
    FROM: <sip:bjtest@ch.com.cn>;tag=aeae9a9e71;epid=6B729D4963
    TO: <sip:13812345678@10.1.1.210;user=phone>;tag=64712-1

    the 'FROM 'will be changed
    sip:bjtest@ch.com.cn -----will be transfered to--------------> sip:+8601012345678@ch.com.cn


    but when I got the log from the gateway that connected to the ocs R2 mediation server, I found the “FROM” is not changed,dose R2 Make the change?





    ---------------------------------------------------
    [01/01 08:10:47.106559]SIP: recv from IP[10.1.1.98:49625],len=[893],msg=
    INVITE sip:13812345678@10.1.1.210;user=phone SIP/2.0
    FROM: <sip:bjtest@ch.com.cn>;epid=6B729D4963;tag=aeae9a9e71
    TO: <sip:13812345678@10.1.1.210;user=phone>
    CSEQ: 1 INVITE
    CALL-ID: 34097713-fdf8-4752-99c2-ee84cc21b0c1
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP 10.1.1.98:49625;branch=z9hG4bKa839fda
    CONTACT: <sip:ocs2.ch.com.cn:5060;transport=Tcp;maddr=10.1.1.98;ms-opaque=cccde79a35a26c86>
    CONTENT-LENGTH: 317
    SUPPORTED: 100rel
    USER-AGENT: RTCC/3.5.0.0 MediationServer
    CONTENT-TYPE: application/sdp; charset=utf-8
    ALLOW: UPDATE
    ALLOW: Ack, Cancel, Bye,Invite

    v=0
    o=- 2 1 IN IP4 10.1.1.98
    s=session
    c=IN IP4 10.1.1.98
    b=CT:1000
    t=0 0
    m=audio 60500 RTP/AVP 97 101 13 0 8
    c=IN IP4 10.1.1.98
    a=rtcp:60501
    a=label:Audio
    a=rtpmap:97 RED/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=rtpmap:13 CN/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=ptime:20

    [01/01 08:10:47.142889]SIP:send TCP ip[10.1.1.98:49625] (len=281/15)=
    SIP/2.0 100 Trying
    Via: SIP/2.0/TCP 10.1.1.98:49625;branch=z9hG4bKa839fda
    To: <sip:13812345678@10.1.1.210;user=phone>;tag=64712-1
    From: <sip:bjtest@ch.com.cn>;tag=aeae9a9e71;epid=6B729D4963
    Call-ID: 34097713-fdf8-4752-99c2-ee84cc21b0c1
    CSeq: 1 INVITE
    Content-Length: 0


    [01/01 08:10:50.162432]SIP:send TCP ip[10.1.1.98:49625] (len=296/15)=
    SIP/2.0 506 Server Internal Error
    Via: SIP/2.0/TCP 10.1.1.98:49625;branch=z9hG4bKa839fda
    To: <sip:13812345678@10.1.1.210;user=phone>;tag=64712-1
    From: <sip:bjtest@ch.com.cn>;tag=aeae9a9e71;epid=6B729D4963
    Call-ID: 34097713-fdf8-4752-99c2-ee84cc21b0c1
    CSeq: 1 INVITE
    Content-Length: 0


    [01/01 08:10:50.182559]SIP: recv from IP[10.1.1.98:49625],len=[327],msg=
    ACK sip:13812345678@10.1.1.210;user=phone SIP/2.0
    FROM: <sip:bjtest@ch.com.cn>;tag=aeae9a9e71;epid=6B729D4963
    TO: <sip:13812345678@10.1.1.210;user=phone>;tag=64712-1
    CSEQ: 1 ACK
    CALL-ID: 34097713-fdf8-4752-99c2-ee84cc21b0c1
    MAX-FORWARDS: 70
    VIA: SIP/2.0/TCP 10.1.1.98:49625;branch=z9hG4bKa839fda
    CONTENT-LENGTH: 0


    Baal
    Thursday, January 28, 2010 1:14 PM

All replies

  • I think when the gateway got the signaling  from mediation server ,the FROM will be changed from “ FROM: <sip:bjtest@ch.com.cn>;tag=aeae9a9e71;epid=6B729D4963 ” to “FROM: <sip:+8601012345678@ch.com.cn>;tag=aeae9a9e71;epid=6B729D4963 ” .


    or

    is there somewhere that can config the behavior?
    Baal
    Thursday, January 28, 2010 1:22 PM
  • I have this isue when invite by phone when in a confernece

    do you find te solution?

     

    thanks

     

    Thursday, October 7, 2010 11:59 AM
  • Same problem only with conference on scenario MOC-PSTN-PSTN, i fixed main issue setting TelephonyMode Policy on Local Computer for fisrt call issue, but second call stils with sip:user@domain

    do you find the solution???

     

    Thank You

    Wednesday, November 3, 2010 6:41 AM