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OCS and CCM6 integration - Problem in incoming calls

Question
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Hi to all,
I've this scenario..
OCS <----> Mediation server <-----> CCM6
The calls started from MOC to CCM phone( or PSTN) works fine. But if I try to make a call from CCM to MOC it doesn't work.
Mediation Server replies with 400 Internal Server Error Bad Information in Contact.
This is the INVITE:
Request-Line: INVITE sip:7765802@10.107.3.193:5060 SIP/2.0
Method: INVITE
[Resent Packet: False]
Message Header
Via: SIP/2.0/TCP 10.107.3.194:5060;branch=z9hG4bK3be701e9c
Remote-Party-ID: "Marino" <sip:1001@10.107.3.194>;party=calling;screen=yes;privacy=off
From: "Marino" <sip:1001@10.107.3.194>;tag=160a0c96-0e59-428b-91c3-13ba19122fcf-31028707
To: <sip:+7765802@10.107.3.193>
Date: Fri, 29 Aug 2008 08:42:55 GMT
Call-ID: 78045700-8b71b68f-56-c2036b0a@10.107.3.194
Supported: timer,replaces
Min-SE: 1800
User-Agent: Cisco-CCM6.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Contact: <sip:1001@10.107.3.194:5060;transport=tcp>
Expires: 180
Allow-Events: presence
Session-Expires: 1800
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 212
Message Body
and this is a cut of ocs debug error...
TL_INFO(TF_PROTOCOL) [0]0B1C.11EC::08/29/2008-08:45:03.531.00016042 (SIPStack,SIPAdminLog::TraceProtocolRecord:1224.idx(122))$$begin_record
Instance-Id: 000000BF
Direction: incoming
Peer: 10.107.3.193:3369
Message-Type: request
Start-Line: INVITE sip:+7765802;phone-context=prova@ocsdemo.it;user=phone SIP/2.0
From: <sip:1001;phone-context=unknown@ocsdemo.it;user=phone>;epid=0A26C2D5B5;tag=7a32f57232
To: <sip:+7765802;phone-context=prova@ocsdemo.it;user=phone>
CSeq: 29 INVITE
Call-ID: d67852f4-b086-436f-9659-e9bf524f70a9
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.107.3.193:3369;branch=z9hG4bK83f84aa2
CONTACT: <sip:mediation.ocsdemo.it@ocsdemo.it;gruu;opaque=srvr:MediationServer:LdsZ1yabAUKNTk9nJE4o6AAA;grid=370cdd169c5940a1a3b21b7bef7e98c0>;isGateway
CONTENT-LENGTH: 929
SUPPORTED: replaces
SUPPORTED: gruu-10
USER-AGENT: RTCC/3.0.0.0 MediationServer
CONTENT-TYPE: application/sdp; charset=utf-8
ALLOW: UPDATE
ms-call-source: non-ms-rtc
ALLOW: Ack, Cancel, Bye,Invite,Refer
Message-Body: v=0
o=- 0 0 IN IP4 10.107.3.193
s=session
c=IN IP4 10.107.3.193
b=CT:1000
t=0 0
m=audio 61183 RTP/AVP 97 101 115 111 0 8
c=IN IP4 10.107.3.193
a=rtcp:60033
a=candidate:JC1W7SuD6x3O06r9jGYogWUt/7bCyE109PHzJEpc8yg 1 pMqNBtRw34oDRimuhk7ZEA UDP 0.900 10.107.3.193 61183
a=candidate:JC1W7SuD6x3O06r9jGYogWUt/7bCyE109PHzJEpc8yg 2 pMqNBtRw34oDRimuhk7ZEA UDP 0.900 10.107.3.193 60033
a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:CXTNYzqCbBmz8d+MbmLxPS1eKZmLCcJz46y+PtAE|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:fYbMZrtD+HpJnIxgm4hLEmCoC3jwzuqGC0OyuHcv|2^31|1:1
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:hjCRVNhF3OxHC9ljyIuEdeufOBuIiYC4wB7HvEEP|2^31
a=label:main-audio
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:115 x-msrta/8000
a=fmtp:115 bitrate=11800
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
$$end_record
TL_ERROR(TF_COMPONENT) [0]0B1C.11EC::08/29/2008-08:45:03.531.0001605c (SIPStack,CContactProcessor:rocessIncomingContactEntry:1280.idx(585))Disallowing server GRUU in message coming from client
TL_WARN(TF_COMPONENT) [0]0B1C.11EC::08/29/2008-08:45:03.531.0001605d (SIPStack,CContactHeader:rocessContactEntries:1069.idx(322))( 031C87E0 ) exit. failed to process contact entry. Returned 0xC3E93F13(SIPPROXY_E_CONTACT_NOT_AUTHORIZED)
TL_WARN(TF_COMPONENT) [0]0B1C.11EC::08/29/2008-08:45:03.531.0001605e (SIPStack,CSIPMessage::_ProcessEachContactEntry:187.idx(979))( 0321BA38 ) exit. failed processing contact entries in contact header Returned 0xC3E93F13(SIPPROXY_E_CONTACT_NOT_AUTHORIZED)
TL_ERROR(TF_COMPONENT) [0]0B1C.11EC::08/29/2008-08:45:03.531.0001605f (SIPStack,CSIPRequest:rocessContactHeaders:311.idx(1653))( 0321BA38 ) Exit - failed to process contact headers Returned 0xC3E93F13(SIPPROXY_E_CONTACT_NOT_AUTHORIZED)
TL_ERROR(TF_COMPONENT) [0]0B1C.11EC::08/29/2008-08:45:03.531.00016060 (SIPStack,CSIPRequest::ValidateInboundHeaders:311.idx(1534))( 0321BA38 ) Exit - Contact parsing failed. Returned 0xC3E93F13(SIPPROXY_E_CONTACT_NOT_AUTHORIZED)
TL_WARN(TF_DIAG) [0]0B1C.11EC::08/29/2008-08:45:03.531.00016063 (SIPStack,SIPAdminLog::TraceDiagRecord:1224.idx(142))$$begin_record
LogType: diagnostic
Severity: warning
Text: Routing error occurred; check Result-Code field for more information
Result-Code: 0xc3e93f13 SIPPROXY_E_CONTACT_NOT_AUTHORIZED
SIP-Start-Line: INVITE sip:7765802;phone-context=prova@ocsdemo.it;user=phone SIP/2.0
SIP-Call-ID: d67852f4-b086-436f-9659-e9bf524f70a9
SIP-CSeq: 29 INVITE
Peer: 10.107.3.193:3369
$$end_record
Someone can help me to solve this issue? Many thanks to all!!Friday, August 29, 2008 8:47 AM
All replies
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Hi,
In your debug, it shows a call going to the following number:
+7765802
Have you assigned that exact number to a user in their line uri property in the communications tab of the user object?
If not, then OCS doesn't know who to route the call to. From the looks of it, that's not the user's full E164 number. You may need a normalization rule for that kind of number. And if that is the case, be sure to assign a location profile to your mediation server so that it will actually use the normalization rules.
regards,
Matt
Tuesday, September 2, 2008 8:15 PM -
You should consider reading this:
http://forums.microsoft.com/unifiedcommunications/ShowPost.aspx?PostID=3779246&SiteID=57
Wednesday, September 3, 2008 6:32 PM