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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

Question
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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?
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[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,Invitev=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
BaalThursday, January 28, 2010 1:14 PM
All replies
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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?
BaalThursday, 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