Hello, I have a working FS instance I am using as a redirect server (to serve up LNP requests). It works fine on calls with invites that have SDP and does not work with invites without SDP. I enabled 3pcc to true thinking that would fix the issue. Version info is FreeSWITCH Version 1.0.6 (hacked-20100921T052029Z). With the console log level set to debug the only thing I see is this message (just before returning a 480): freeswitch@lnpdal0001> 2010-11-10 08:58:50.949818 [CRIT] switch_core_state_machine.c:382 e390f5b0-ecda-11df-906a-13020c8dafd9 Timeout waiting for next instruction in CS_NEW! I also do not ever see FS to an xml http req (which is how we control the dialplan). freeswitch@lnpdal0001> recv 495 bytes from udp/[X.X.X.X]:5060 at 14:57:50.934431: ------------------------------------------------------------------------ INVITE sip:19033226103@X.X.X.X:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 Max-Forwards: 70 Contact: <sip:X.X.X.X:5060> To: <sip:19033226103@X.X.X.X:5060> From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE Date: Tue, 09 Nov 2010 15:29:25 GMT Supported: 100rel User-Agent: ENSR3.0.63.0-IS1-RMRG2101-RG21-CPO11152 Content-Length: 0 ------------------------------------------------------------------------ send 336 bytes to udp/[X.X.X.X]:5060 at 14:57:50.935324: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060> Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Content-Length: 0 ------------------------------------------------------------------------ send 714 bytes to udp/[X.X.X.X]:5060 at 14:58:50.951429: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK11288997ec8ll7466dfINV1db35e674cd968d5 From: <sip:X.X.X.X:5060>;tag=1db35e67-co6935-INS001 To: <sip:19033226103@X.X.X.X:5060>;tag=t7ycB64mX33mQ Call-ID: 993186-34928005397-80464@ens.com CSeq: 693501 INVITE User-Agent: lnpdal0001.sipinterchange.com Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=81;text="INVALID_CALL_REFERENCE" Content-Length: 0 Remote-Party-ID: "19033226103" <sip:19033226103@X.X.X.X
;party=calling;privacy=off;screen=no
------------------------------------------------------------------------ Here is the sofia profile: <profile name="external"> <aliases> <alias name="outbound"/> <alias name="nat"/> <!-- for backwards compatibility --> </aliases> <domains> <domain name="all" alias="false" parse="true"/> </domains> <settings> <param name="user-agent-string" value="lnpdal0001"/> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="context" value="public"/> <param name="enable-100rel" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="sip-ip" value="X.X.X.X"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${external_ssl_enable}"/> <!-- <param name="pass-rfc2833" value="true"/> --> <!-- <param name="inbound-proxy-media" value="true"/> --> <param name="inbound-bypass-media" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="accept-blind-reg" value="false"/> <param name="accept-blind-auth" value="true"/> <param name="nonce-ttl" value="60"/> <param name="disable-transcoding" value="true"/> <param name="auth-calls" value="false"/> <param name="inbound-reg-force-matching-username" value="false"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="enable-3pcc" value="true"/> </settings> </profile> Any thoughts / ideas / help would be greatly appreciated. Thanks! SDR