Steve, Thanks for the assistance. The entire console log after setting console loglevel 9 is as follows: 2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382 f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in CS_NEW! Everything else is just the SIP messaging. It's like FS never gets the call from sofia for some reason. I tried the following (your comments around the media make sense for sure): -- remove the disable transcoding (this did not work) -- remove the other codec parameters (this did not work) To clarify what the box does: -- takes an invite -- responds with a 302 redirect with a modified contact header that looks like $dn;npdi=yes;rn=$rn (if there is an rn)@$host info. This is done with using mod_xml_curl to provide the dynamic dialplan. There are not ever any b legs. Every call is an invite, 100 trying, 302, ACK. In this case we get an invite, a 100 trying and that's it. Doing an ngrep -d any -qW byline port 80 shows that there is not a xml_curl req and on the console at loglevel 9 the only message from FS is this: 2010-11-10 10:11:02.501493 [CRIT] switch_core_state_machine.c:382 f96f23a2-ece4-11df-9888-e3878ca86765 Timeout waiting for next instruction in CS_NEW! Sofia spits out all of the SIP messaging but that's it. It looks nothing like a normal call when I send the SDP. It's a little odd. Here is the modified profile after commenting out all of the info: <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.sipinterchange.com"/> <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="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="4.71.122.205"/> <param name="sip-ip" value="4.71.122.205"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="bind-params" value="transport=udp"/> <param name="tls" value="$${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> Thanks again for the help. SDR