Mediation Server -> Asterisk/SER
Question
-
Hi,
I'd like to ask for a little help.
Hope you guys can help me fix this problem.
Its about using Asterisk as Media Gateway.
(OCS client --> OCS Server --> tls --> Mediation Server --> tcp --> SER
--> udp--> Asterisk --> sip-phone)
The SER configuration is ok .
But i cant still create any call.
From Asterisk to OCS client the INVITE request is ok but the mediation server is allways cancel the call.
S4.log // looks ok
<snippet>
>>> Incoming TCP packet BEGIN
INVITE sip:+49XXXXX1@domain.local SIP/2.0\r\n
Record-Route: <sip:192.168.100.55;ftag=as3700c801;lr=on>\r\n
Via: SIP/2.0/TCP
192.168.100.55;branch=z9hG4bKa39d.46a7c0622426a6ada79426342d42cb3b.0\r\n
Via: SIP/2.0/UDP 192.168.100.51:5060;branch=z9hG4bK7e46cd43;rport=5060\r\n
From: ":+49XXXXX2" <sip:+49XXXXX2@192.168.100.51>;tag=as3700c801\r\n
To: <sip:+49XXXXX1@domain.local>\r\n
Contact: <sip:+49XXXXX2@192.168.100.51>\r\n
Call-ID: 1f84eee54ebff5ae41116d3408552a7f@192.168.100.51\r\n
CSeq: 102 INVITE\r\n
User-Agent: Asterisk PBX\r\n
Max-Forwards: 16\r\n Date: Thu, 16 Aug 2007 08:39:18 GMT\r\n
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY\r\n
Content-Type: application/sdp\r\n
Content-Length: 242\r\n \r\n v=0\r\n o=root 3995 3995 IN IP4 192.168.100.51\r\n s=session\r\n c=IN IP4 192.168.100.51\r\n t=0 0\r\n m=audio 60614 RTP/AVP 0 8 101\r\n a=rtpmap:0 PCMU/8000\r\n
a=rtpmap:8 PCMA/8000\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=fmtp:101 0-16\r\n
a=silenceSupp
ff - - - -\r\n<<< Incoming TCP packet END
</snippet>
Found in mediationserver.txt:
<snippet>
....
$$END-MEDIATIONSERVER
TL_ERROR(TF_COMPONENT) [0]0ED4.0EDC::08/16/2007-08:33:42.891.00000003
(MediationServer,GatewayListener.GatewaySessionReceivedEventHandler:854.idx(190))(
02E4C884 )Exception: The header parsing failed.
</snippet>
And more details in collaboration.txt:
<snippet>
> Message: Invalid RecordRoute header: parsing problem (no viable
alternative): =
> TargetSite: Void ParseRecordRouteHeaders()
> StackTrace:
at Microsoft.Rtc.Internal.Sip.SipMessageBase.ParseRecordRouteHeaders()
at Microsoft.Rtc.Internal.Sip.SipMessageBase.get_RecordRouteHeaders()
at
Microsoft.Rtc.Signaling.SignalingSession.BuildSignalingHeaders(SipMessage
sipMessage) > Source: SIPEPS
.....
// cancel call request will be created here ....
....
L_WARN(TF_COMPONENT) [0]0AD4.0B4C::08/19/2007-08:42:49.513.00000014
(Collaboration,RealTimeConnectionManager.CoreManagerIncomingCancelTransactionCreated:519.idx(2347))(
038733AB )<RealTimeServerTcpConnectionManager_38733AB> Exception:
Exception: Microsoft.Rtc.Internal.Sip.SipException
> Exception: Exception: Microsoft.Rtc.Internal.Sip.SipException
> Message: Invalid RecordRoute header: parsing problem (no viable
alternative): =
> TargetSite: Void ParseRecordRouteHeaders() > StackTrace:
at Microsoft.Rtc.Internal.Sip.SipMessageBase.ParseRecordRouteHeaders()
at Microsoft.Rtc.Internal.Sip.SipMessageBase.get_RecordRouteHeaders()
at Microsoft.Rtc.Internal.Sip.SipResponse.SipResponseHelper(Int32
status, SipRequest request)
at Microsoft.Rtc.Internal.Sip.SipResponse..ctor(Int32 status, SipRequest
request)
at Microsoft.Rtc.Signaling.SignalingSession.CreateSipResponse(SipRequest
request, Int32 responseCode, IEnumerable`1 signalingHeaders)
at
Microsoft.Rtc.Signaling.SignalingSession.HandleCancelRequest(SipTransaction
sipTransaction)
at
Microsoft.Rtc.Signaling.SipSignalingSessionManager.CoreManagerIncomingCancelTransactionCreated(Object
sender, TransactionCreatedEventArgs e) at Microsoft.Rtc.Signaling.RealTimeConnectionManager.CoreManagerIncomingCancelTransactionCreated(Object
sender, TransactionCreatedEventArgs e)
> Source: SIPEPS
</snippet>
Invalid RecordRoute header ?? Looks ok for me....
thanks
