Event Type:	Warning
Event Source:	Microsoft Exchange Speech Engine
Event Category:	Telephony Application Host 
Event ID:	32768
Date:		2/2/2007
Time:		6:29:31 PM
User:		N/A
Computer:	SWORDFISH
Description:
The Telephony Manager declined a call with Call Id '4a639f776f41526427ccdca11a7f6f22@66.192.107.225' for the following reason in component speech session: 'Transport error occurred while sending.'.

Further trace information for support personnel follows: 
Microsoft.Collaboration.SignalingException: Transport error occurred while sending. ---> Microsoft.Sip.SipException: Invalid RecordRoute header
   at Microsoft.Sip.SipMessageBase.ParseRecordRouteHeaders()
   at Microsoft.Sip.SipMessageBase.get_RecordRouteHeaders()
   at Microsoft.Sip.SipResponse.SipResponseHelper(Int32 status, SipRequest request)
   at Microsoft.Collaboration.Signaling.SipSignalingSession.CreateSipResponse(SipTransaction transaction, Int32 responseCode, IEnumerable`1 signalingHeaders)
   at Microsoft.Collaboration.Signaling.SipSignalingSession.SendProvisionalResponse(Int32 responseCode, IEnumerable`1 signalingHeaders)
   --- End of inner exception stack trace ---
   at Microsoft.Collaboration.Signaling.SipSignalingSession.SendProvisionalResponse(Int32 responseCode, IEnumerable`1 signalingHeaders)
   at Microsoft.SpeechServer.Core.TelephonyManager.SignalingSessionReceived(Object sender, SessionReceivedEventArgs e), Response Code: 0

For more information, see Help and Support Center at http://go.microsoft.com/fwlink/events.asp.





####
T +11.784064 openser:32878 -> swordfish:5065 [AP]
INVITE sip:9090@openser SIP/2.0.
Record-Route: <sip:openser;transport=tcp;r2=on;lr=on;ftag=as2e930dc4>.
Record-Route: <sip:openser;r2=on;lr=on;ftag=as2e930dc4>.
Via: SIP/2.0/TCP openser;branch=z9hG4bKd9c8.c2680e24.0.
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK585e25e3;rport=5060.
From: "Jon Webster" <sip:3149@asterisk>;tag=as2e930dc4.
To: <sip:8889090@openser>.
Contact: <sip:3149@asterisk>.
Call-ID: 4a639f776f41526427ccdca11a7f6f22@asterisk.
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 69.
Date: Fri, 02 Feb 2007 23:29:43 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Content-Type: application/sdp.
Content-Length: 630.
.
v=0.
o=root 381 381 IN IP4 asterisk.
s=session.
c=IN IP4 asterisk.
t=0 0.
m=audio 19734 RTP/AVP 0 3 4 8 111 5 10 7 18 110 97 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:111 G726-32/8000.
a=rtpmap:5 DVI4/8000.
a=rtpmap:10 L16/8000.
a=rtpmap:7 LPC/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:110 speex/8000.
a=rtpmap:97 iLBC/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
m=video 17590 RTP/AVP 34 26 31 103.
a=rtpmap:34 H263/90000.
a=rtpmap:26 JPEG/90000.
a=rtpmap:31 H261/90000.
a=rtpmap:103 h263-1998/90000.

#
T +0.000902 swordfish:5065 -> openser:32878 [AP]
SIP/2.0 100 Trying.
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as2e930dc4.
TO: <sip:8889090@openser>.
CSEQ: 102 INVITE.
CALL-ID: 4a639f776f41526427ccdca11a7f6f22@asterisk.
MAX-FORWARDS: 70.
VIA: SIP/2.0/TCP openser;branch=z9hG4bKd9c8.c2680e24.0,SIP/2.0/UDP asterisk:5060;branch=z9hG4bK585e25e3;rport=5060.
CONTENT-LENGTH: 0.
.

##
T +0.003498 swordfish:5065 -> openser:32878 [AP]
SIP/2.0 500 Internal Server Error.
FROM: "Jon Webster"<sip:3149@asterisk>;tag=as2e930dc4.
TO: <sip:8889090@openser>;tag=c072ca161a.
CSEQ: 102 INVITE.
CALL-ID: 4a639f776f41526427ccdca11a7f6f22@asterisk.
MAX-FORWARDS: 70.
VIA: SIP/2.0/TCP openser;branch=z9hG4bKd9c8.c2680e24.0,SIP/2.0/UDP asterisk:5060;branch=z9hG4bK585e25e3;rport=5060.
CONTENT-LENGTH: 0.
SERVER: RTCC/2.0.6017.0.
.

##
T +0.000563 openser:32878 -> swordfish:5065 [AP]
ACK sip:9090@openser SIP/2.0.
Via: SIP/2.0/TCP openser;branch=z9hG4bKd9c8.c2680e24.0.
From: "Jon Webster" <sip:3149@asterisk>;tag=as2e930dc4.
Call-ID: 4a639f776f41526427ccdca11a7f6f22@asterisk.
TO: <sip:8889090@openser>;tag=c072ca161a.
CSeq: 102 ACK.
User-Agent: OpenSer (1.1.1-tls (i386/linux)).
Content-Length: 0.
.

###